| 00:01.23 | Janos | I have a TE121 and a AEX800 in the same system. Which according to digium page could come with a VPMOCT032, so assuming this is true can anyone tell me which is the firmware for this EC ? |
| 00:02.07 | *** join/#asterisk JAMMAN2110 (~JAMMAN211@unaffiliated/jamman2110) |
| 00:03.33 | carrar | Janos, like the docs in digium suport? |
| 00:03.34 | carrar | http://kb.digium.com/articles/FAQ/Is-my-VPMADT032-installed-and-operational?retURL=%2Fapex%2FknowledgeHome&popup=false |
| 00:03.37 | carrar | http://kb.digium.com/articles/FAQ/How-do-I-disable-echo-cancellation-module-without-removing-the-hardware?retURL=%2Fapex%2FknowledgeHome&popup=false |
| 00:04.21 | carrar | grep for VPM |
| 00:10.19 | Janos | carrar, thanks a lot for the docs, but i´m pretty sure VPM is not loading since i´m getting Failed to load the firmware in dmesg, now what i´m trying to figure out here is, which is the right firmware and why is not been loaded, i installed a package called dahdi-firmware-nonfree in debian which gave me three files in /lib/firmware/{dahdi-fw-oct6114-064.bin,dahdi-fw-oct6114-128.bin,dahdi-fw-tc400m.bin} |
| 00:12.45 | WIMPy | Do you have udev running? |
| 00:13.16 | Janos | WIMPy, yes |
| 00:14.50 | ruben23 | hi guys..anyone have link for howto or any idea on impelementing fail2ban on asterisk |
| 00:15.46 | Janos | maybe there firmware needs to be present when i build the modules in order to work ? |
| 00:15.59 | Janos | cuz i know i wasn´t |
| 00:16.22 | carrar | ruben32, you did google "How to install fail2ban with asterisk" ? |
| 00:17.10 | carrar | Janos, might want to call Digium for support on their hardware |
| 00:17.34 | carrar | Though you will probably need the serial number off the card as well |
| 00:19.13 | Janos | carrar, kk thanks |
| 00:28.56 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 00:37.07 | dijib | glorious ringing |
| 00:38.46 | *** join/#asterisk tzafrir (~tzafrir@212.179.75.202) |
| 00:39.03 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
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| 00:58.44 | *** join/#asterisk CoderForLife (~Miranda@unaffiliated/coderforlife) |
| 01:10.06 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 01:13.47 | SeRi | p3nguin, you in? |
| 01:14.04 | SeRi | I figure out why is the gateway is sending the number as name |
| 01:14.54 | SeRi | because is set to Tel/User and that means cid user@asterisk |
| 01:15.44 | p3nguin | I'm not concerned with the number in the name field; I'd rather know how to change it so that it sends <sip:CallerIDnumber@address> |
| 01:17.58 | *** join/#asterisk JuStIcIa_ (~JuStIcIa_@190.80.138.97) |
| 01:18.14 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 01:18.14 | SeRi | tel/tel |
| 01:18.23 | SeRi | Thats how the gsm does it |
| 01:18.42 | SeRi | tel tel will do that sip:cid@asterisk |
| 01:19.04 | p3nguin | So if you do Tel/Tel, do you get From: CallerName <sip:callerNumber@address> ? |
| 01:19.38 | SeRi | and thats when I had issues before where the device wanted to force a registartion usingg cid@asterisk and since tahts not valid it wouldnt work |
| 01:19.57 | SeRi | p3nguin, I get cid sip:cid@asterisk |
| 01:20.02 | p3nguin | CallerID has nothing to do with registration. |
| 01:20.15 | SeRi | I know thats how bad thise device is about it |
| 01:20.24 | SeRi | is like is built backwards |
| 01:20.47 | p3nguin | For the device to register, there is NO caller ID information. CallerID has nothing to do with registration. |
| 01:20.55 | SeRi | one sec |
| 01:21.18 | p3nguin | You're mixing registration with calling. They are not related. |
| 01:22.16 | p3nguin | If you can get calls to come through as <sip:callerIDnumber@host>, that's all you need to do. That's all I wanted to do earlier. |
| 01:29.04 | *** join/#asterisk adeel (~adeel@24-246-63-106.cable.teksavvy.com) |
| 01:29.18 | *** join/#asterisk Hanumaan (~Hanumaan@132.230.17.77) |
| 01:29.59 | SeRi | ok. |
| 01:30.09 | SeRi | Thanks for the help |
| 01:30.46 | p3nguin | Do calls come through with the real callerID number now? |
| 01:32.08 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net) |
| 01:32.22 | KingDavidNYC | Hello everybody |
| 01:32.29 | p3nguin | I'm looking back at the manual for your device. |
| 01:32.38 | SeRi | p3nguin, ok |
| 01:32.40 | p3nguin | I see where it says the caller number comes through as the name. |
| 01:33.32 | SeRi | :) does it make sense? |
| 01:33.35 | p3nguin | User/User is completely out of the question. |
| 01:33.43 | SeRi | Yes |
| 01:33.47 | p3nguin | Yes, it makes sense now that I've talked to you about it a little. |
| 01:33.52 | SeRi | right now is Tel/User |
| 01:34.03 | p3nguin | I don't necessarily agree with the way they do it, but it makes sense. |
| 01:34.23 | SeRi | pe nether. I think is technically broken :/ |
| 01:34.27 | SeRi | me* |
| 01:34.32 | SeRi | but it works |
| 01:34.35 | SeRi | I gues.. |
| 01:34.37 | SeRi | lol |
| 01:34.38 | SeRi | :) |
| 01:35.00 | p3nguin | Tel/Tel would be okay, but that doesn't allow registrations. |
| 01:35.15 | SeRi | exactly. BUT! I think I found a bug. |
| 01:35.41 | SeRi | it does try to send a registration in as the cid@the device |
| 01:35.46 | KingDavidNYC | Can anyone please help me on my first time Realtime setup?, the phone is not registering :( |
| 01:35.51 | p3nguin | It can't do that. |
| 01:36.00 | p3nguin | Caller ID is for CALLS. |
| 01:36.11 | p3nguin | I don't know how else I can say it. |
| 01:36.23 | p3nguin | There is no caller ID on a REGISTER. |
| 01:37.09 | SeRi | ok I guess I was looking at it wrong when I was seen the errors. |
| 01:37.18 | p3nguin | Maybe you were looking at an INVITE. |
| 01:37.35 | p3nguin | Or maybe the REGISTER had something else that you were getting confused with a Caller ID number. |
| 01:37.37 | SeRi | probably. |
| 01:37.49 | SeRi | KingDavidNYC, please post your error |
| 01:37.50 | p3nguin | Register is the way a device tells asterisk "Here I am." |
| 01:38.01 | p3nguin | "This is how you can send me calls." |
| 01:38.05 | p3nguin | That's all it does. |
| 01:38.20 | SeRi | I see |
| 01:38.36 | p3nguin | If you look at a REGISTER packet, you'll see a "Contact" field. |
| 01:39.09 | KingDavidNYC | config.c:2018 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available |
| 01:39.12 | p3nguin | Now... if I had the device, I would try setting it up on Tel/Tel. |
| 01:39.20 | KingDavidNYC | Seri: tahnks |
| 01:39.43 | KingDavidNYC | ver 1.6.20 |
| 01:39.44 | p3nguin | It won't register, so we'll treat it like an ITSP. You'll change host=dynamic to host=ip-address-of-the-device |
| 01:40.08 | SeRi | p3nguin, Thats what I wanted to do before |
| 01:40.19 | p3nguin | It will still be able to accept calls from asterisk and it will still be able to send calls in. |
| 01:40.40 | SeRi | Thats what I wanted to do. thats my ultimate goal. |
| 01:40.49 | p3nguin | Since the manual says calls will come from "caller number" <sip:caller number@host>, I'd be fine with that. |
| 01:41.18 | SeRi | KingDavidNYC, you have mysql enable? |
| 01:41.31 | p3nguin | Switch it over to Tel/Tel. Configure the proxy server IP for asterisk's IP address, and set Active to On. |
| 01:41.45 | KingDavidNYC | Seri: yes, all is setup |
| 01:41.48 | SeRi | ok |
| 01:41.54 | p3nguin | Change the host= entry in sip.conf to be the device's IP address. |
| 01:42.07 | SeRi | p3nguin, ok. |
| 01:42.14 | KingDavidNYC | Seri: table sipfriends exists |
| 01:42.49 | SeRi | KingDavidNYC, I am not an expert by a long shot but asterisk is complaining that mysql is not working properly. |
| 01:42.55 | KingDavidNYC | Seri: the table exists, the db connects, extconfig is set, and res_mysql.conf also |
| 01:43.01 | SeRi | p3nguin, I am working on it |
| 01:43.24 | KingDavidNYC | Seri: mysql is working |
| 01:44.01 | KingDavidNYC | Seri: What probably is not doing is being able to read the table in one of the configuration files |
| 01:44.29 | *** join/#asterisk adeel (~adeel@24.246.63.106) |
| 01:44.31 | KingDavidNYC | Seri: do I have to set any rtXXX in sip.conf |
| 01:44.38 | SeRi | KingDavidNYC, some of the vets here can help more with that than I can. I am still a n00b. |
| 01:45.22 | SeRi | p3nguin, so erase all the registartion info and just put in the proxy and leave the rest empty? |
| 01:46.33 | p3nguin | I can't see what it looks like, but it says don't register. It says only fill 'proxy server IP' and choose 'Acitve: on' (else field empty). |
| 01:47.00 | p3nguin | I guess else field empty means everything that isn't proxy server IP and Active needs to be empty. |
| 01:47.05 | SeRi | It looks just like img in the manual. ok trying it now. |
| 01:47.18 | p3nguin | I don't speak Taiwanese. |
| 01:47.36 | KingDavidNYC | Seri: Thank you Seri, I thought that since you asked me to show you the problem, you would know |
| 01:48.46 | SeRi | KingDavidNYC, I can help with very little simple stuff. Just wanted you to post the issue so people can see. Usually is how it goes... :) |
| 01:49.05 | SeRi | p3nguin, Yea the manual is so damn broken... ugh... :( |
| 01:50.26 | SeRi | p3nguin, so I take secret out of the sip.conf for that peer right |
| 01:50.36 | p3nguin | no |
| 01:51.08 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 01:51.27 | SeRi | ok |
| 01:51.34 | SeRi | took of dynamic and set the ip than |
| 01:51.45 | p3nguin | I can't see the device config interface, so I don't know how it will authenticate stuff. |
| 01:53.26 | SeRi | well if it wont. tel/tel uses no authentication |
| 01:53.41 | SeRi | well it wont* |
| 01:54.12 | p3nguin | Take off the secret if you want. That's how testing works, right? |
| 01:54.34 | SeRi | :) |
| 01:54.40 | p3nguin | I found a picture of the page it was talking about. |
| 01:55.10 | p3nguin | I guess it wants all the fields empty except Proxy Server. |
| 01:55.25 | SeRi | <PROTECTED> |
| 01:55.44 | SeRi | you see it trys to force registration. the device is set NOT to register |
| 01:55.49 | SeRi | set* |
| 01:55.54 | p3nguin | That's silly. |
| 01:56.06 | *** join/#asterisk luckman212_phone (~luckman21@cpe-24-193-125-224.nyc.res.rr.com) |
| 01:56.28 | SeRi | let me try a call and see what happens |
| 01:56.49 | p3nguin | If you take out display name, user name, register name, register password... submit it, then reboot the device, I bet you won't see that again. |
| 01:57.29 | SeRi | Its all like that all ready :) |
| 01:57.44 | SeRi | BUT. It actually works |
| 01:57.53 | SeRi | just like you said. |
| 01:57.59 | SeRi | tel/tel |
| 01:58.02 | p3nguin | And after everything is empty and you save and reboot it, do you see the register attempt anymore? |
| 01:58.14 | SeRi | Yes. |
| 01:58.22 | p3nguin | That's broken. |
| 01:58.33 | SeRi | Status: Not Registered |
| 01:58.36 | SeRi | lol |
| 01:58.38 | SeRi | It is broken! |
| 01:58.44 | p3nguin | If you blanked all the fields, how could it have a register name to send? |
| 01:58.49 | SeRi | I said that 200000000 lines a go |
| 01:59.28 | SeRi | Well it tried only once. |
| 01:59.34 | SeRi | it did not try to register any more |
| 01:59.39 | p3nguin | Maybe it won't ever try again. |
| 01:59.41 | SeRi | only during reboot |
| 01:59.49 | SeRi | I am hoping |
| 02:00.03 | p3nguin | That'll be okay, I suppose, but it still does not explain how it has a register name if you erased it. |
| 02:00.18 | SeRi | p3nguin, I spoke to soon |
| 02:00.47 | SeRi | look |
| 02:00.53 | SeRi | chan_sip.c:24331 in handle_request_register: Registration from '<sip:MYCELLCID@10.30.2.53>' failed for '10.30.2.57:5060' - No matching peer found |
| 02:01.05 | p3nguin | Was that when you called it? |
| 02:01.11 | SeRi | yes |
| 02:01.17 | p3nguin | hahaha that's stupid. |
| 02:01.28 | SeRi | YES YS YES! |
| 02:01.32 | p3nguin | Trying to send a REGISTER when it needs to send an INVITE... |
| 02:01.36 | p3nguin | stoooopid |
| 02:02.01 | SeRi | broken |
| 02:02.05 | SeRi | lol |
| 02:02.06 | *** join/#asterisk xpot (~xpot@166-70-100-198.ip.xmission.com) |
| 02:02.17 | p3nguin | If it's going to do that, you're not going to be able to use Tel/Tel with no registrations. |
| 02:03.12 | p3nguin | You could use Tel/User, make it register, and rewrite the caller ID number from the caller ID name in its extension. |
| 02:03.40 | p3nguin | Set(CALLERID(num)=${CALLERID(name)}) |
| 02:03.57 | SeRi | Yes sr. |
| 02:04.00 | p3nguin | This is what I call a workaround for a broken device. |
| 02:04.06 | SeRi | agreed |
| 02:04.28 | p3nguin | I don't like workarounds, but I also don't like to spend more money to buy another device. |
| 02:04.40 | SeRi | amen |
| 02:05.01 | p3nguin | And they have defined the From name will always be the caller number... |
| 02:05.08 | p3nguin | so the workaround seems legit. |
| 02:05.35 | p3nguin | This concludes our test. |
| 02:05.43 | SeRi | Thank You se. |
| 02:05.44 | p3nguin | Now you know how you have to configure the broken thing. |
| 02:05.47 | SeRi | sr* |
| 02:09.27 | SeRi | p3nguin, lol |
| 02:23.22 | SeRi | adding more mem to the firewall. |
| 02:23.25 | SeRi | brb |
| 02:30.11 | *** join/#asterisk Zfield (~z@CPE00134936dbc6-CM0011ae922d14.cpe.net.cable.rogers.com) |
| 02:32.10 | Zfield | I'm trying to get asterisk connected to a device that has FXS and FXO ports so I can use my computer as a VoIP gateway to my home phone line to make calls |
| 02:32.33 | Zfield | but I have no clue where to start in terms of settings. the device i have is a d-link dvg-1120S |
| 02:33.07 | Zfield | is anyone able to give me a hand? I'm not sure what settings to set on the device and on asterisknow to get this working |
| 02:34.55 | Zfield | asterisk is setup with a voip phone connected to it sucesfully |
| 02:38.24 | spditner | wonderworld: re: ConfBridge do you autoload all modules, or load specific ones? |
| 02:45.25 | *** join/#asterisk SeRi (~seriosly@c-76-31-169-54.hsd1.tx.comcast.net) |
| 02:45.33 | SeRi | done :) |
| 02:45.49 | SeRi | 4Gigs now :) |
| 02:46.10 | KingDavidNYC | Hi, can anybody please help me figure out why I can not register a phone using realtime in 1.6.2? |
| 02:46.23 | *** join/#asterisk wesphillips (~wphill04@adsl-75-53-137-67.dsl.hstntx.sbcglobal.net) |
| 02:48.21 | *** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net) |
| 02:49.17 | spditner | wonderworld: apparently the minimal config is [default_user]\ntype=user\n[default_bridge]\ntype=bridge\n |
| 02:51.54 | *** part/#asterisk wesphillips (~wphill04@adsl-75-53-137-67.dsl.hstntx.sbcglobal.net) |
| 03:12.43 | p3nguin | 4G? I have 256M, and I'm using about 50M of it. |
| 03:17.46 | *** join/#asterisk gajini (~root@61.12.17.170) |
| 03:17.47 | *** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net) |
| 03:21.38 | SeRi | p3nguin, My firewall runs several modules |
| 03:21.41 | SeRi | I need the ram. |
| 03:21.51 | SeRi | including snort ids which is a mem hog |
| 03:22.35 | SeRi | in to of that I run vnstat2, ntop, squid, unbound, etc.... |
| 03:22.40 | SeRi | top* |
| 03:22.50 | p3nguin | -/+ buffers/cache: 57 183 |
| 03:23.10 | SeRi | nice :) |
| 03:23.19 | SeRi | I been checking out vyatta |
| 03:23.26 | p3nguin | That's what I'm using. |
| 03:24.59 | SeRi | nice |
| 03:27.23 | SeRi | p3nguin, is there a free version of vayatta? |
| 03:27.32 | p3nguin | Yes. |
| 03:27.50 | SeRi | all I see is payed for in there site |
| 03:28.26 | p3nguin | http://vyatta.org/downloads |
| 03:28.54 | *** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net) |
| 03:29.22 | SeRi | o .org i see |
| 03:30.10 | SeRi | going to give it a try. |
| 03:30.16 | SeRi | tired of pfsense bugs |
| 03:31.12 | p3nguin | I like Vyatta quite a bit. |
| 03:31.25 | SeRi | can it keep track of bandwith? |
| 03:32.53 | p3nguin | I guess that depends on exactly what you want to see. |
| 03:33.14 | SeRi | total bandwidth usage per month. I have to track it since I have a cap :( |
| 03:33.50 | SeRi | did you had to forward ports for sip? |
| 03:34.06 | p3nguin | http://pastebin.com/hDTwRz2i |
| 03:35.00 | SeRi | cool |
| 03:35.17 | p3nguin | I have to use NAT rules for SIP, yes. |
| 03:35.40 | SeRi | I now have a week end project! :) |
| 03:35.43 | SeRi | ok cool |
| 03:37.28 | p3nguin | If you are familiar with Cisco, you can probably find your way around Vyatta. There are lots of similarities between the two. |
| 03:38.01 | SeRi | not really but that does not mean I cant figure it out |
| 03:40.10 | p3nguin | I finally decided to set up traffic shaping for voice on my Vyatta system last night. It works very well. |
| 03:43.23 | SeRi | cool |
| 03:44.21 | SeRi | I laso have it setup in pfsense though it works very well its some what buggy and obscure |
| 03:45.54 | *** join/#asterisk master_of_master (~master_of@p57B52975.dip.t-dialin.net) |
| 03:54.49 | SeRi | p3nguin, how did it go with the vm file issue? |
| 03:54.54 | SeRi | you removed it? |
| 03:55.17 | p3nguin | That file that was over 24 hours long? |
| 03:55.23 | SeRi | yes |
| 03:55.55 | p3nguin | I just deleted that file and then set the max voice mail time to 180 seconds so it never happens again. |
| 03:59.31 | SeRi | nice |
| 03:59.44 | SeRi | that was insane! |
| 04:00.51 | p3nguin | I don't really know how it happened, but the message started out with some voice and then it switched to 24 hours of MoH. |
| 04:01.10 | SeRi | a hung channel maybe? |
| 04:06.31 | *** join/#asterisk corretico (~luis@201.201.44.82) |
| 04:12.32 | SeRi | dijib's confference is up. I thought it was down... didnt he had issues earlier? |
| 04:13.13 | SeRi | I guess he got it fixed. |
| 04:15.00 | *** join/#asterisk LiuYan (~liu.yan@211.154.128.135) |
| 04:20.30 | Kobaz | hmm |
| 04:20.40 | Kobaz | is there anything special you need to do to get aastra phones to register |
| 04:20.56 | Kobaz | i seem to have everything configured but the phone doesn't even attempt to register at all |
| 04:28.36 | *** join/#asterisk radic (~radic@dslb-178-002-208-131.pools.arcor-ip.net) |
| 04:29.33 | *** join/#asterisk KingDavidNYC (~Chris1232@pool-74-96-172-188.washdc.fios.verizon.net) |
| 04:32.57 | Kobaz | oh hmm, apparently you need to also fill in screen name and display name |
| 04:39.09 | SeRi | Kobaz, I dont think you need screen name |
| 04:39.27 | SeRi | well actually I think in aastras you do |
| 04:39.31 | SeRi | I take that back |
| 04:42.21 | *** join/#asterisk nafg_ (~quassel@ool-4355e4a2.dyn.optonline.net) |
| 04:46.51 | Kobaz | i dunno |
| 04:46.58 | Kobaz | something it didn't like |
| 04:47.05 | Kobaz | i filled out every field and then it finally worked |
| 04:47.21 | p3nguin | Start taking out one by one and see when it stops working. |
| 04:48.06 | p3nguin | If you take out one and it stops, put it back in and try another. |
| 04:48.20 | *** join/#asterisk tyrok (~tyrok@c-68-41-70-233.hsd1.mi.comcast.net) |
| 04:50.22 | tyrok | Hi! I'm very very new to Asterisk and am running into a problem. The server's set up as basically a bidirectional SIP proxy, and I'm testing it by having it answer, wait 5 seconds, and play back a message to the caller. The problem is, when the caller calls in, it answers, gets through the wait, and drops the call when it gets to the speech part. Considering it's entirely possible I'm messing up something really basic, any ideas? |
| 04:52.13 | tyrok | I can adjust the wait just fine, and that changes how long (from the client end) it takes before the call's dropped, so the extension is running. |
| 04:54.37 | *** part/#asterisk gajini (~root@61.12.17.170) |
| 04:54.50 | *** join/#asterisk jetlag (~jetlag@pool-71-188-5-107.cmdnnj.east.verizon.net) |
| 04:55.03 | nafg_ | Is there any way, when recording a file, to tell asterisk not to accept plain silence? |
| 04:55.53 | p3nguin | tyrok: Try something simple like exten => 123,1,Playback(silence/5&tt-monkeys) where 123 is whatever extension you want to use. |
| 04:57.02 | tyrok | p3nguin: I don't have asterisk-sounds installed currently. |
| 04:57.26 | tyrok | I've been trying using "exten => s,4,Festival("This is a test")" |
| 04:57.33 | p3nguin | None? You deselected the sounds selected by default? |
| 04:58.01 | tyrok | p3nguin: No, but when I looked in the sounds directory, it was empty. |
| 04:58.18 | p3nguin | How did you install asterisk? |
| 04:58.28 | tyrok | yum install asterisk |
| 04:58.34 | tyrok | To give you an idea, Playback(beep) doesn't work. |
| 04:58.38 | p3nguin | Oh, I see. |
| 04:59.14 | tyrok | I could install the DAG package or just unzip them, but as far as I can tell, the problem is due to sending *any* audio. |
| 04:59.46 | tyrok | Festival seems to be synthesizing just fine as far as I can tell. But as soon as it actually tries to send that audio back to the client, it disconnects. |
| 04:59.51 | p3nguin | How about Ringing() or Congestion()? |
| 05:00.19 | tyrok | checks... |
| 05:01.17 | p3nguin | Or even Echo() |
| 05:02.00 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-xbfoknmqqvquultc) |
| 05:03.19 | tyrok | Ringing() seems to work. |
| 05:03.45 | p3nguin | So there's audio. |
| 05:04.08 | tyrok | Let me try PlayTones a minute... |
| 05:04.18 | p3nguin | or Echo() |
| 05:04.44 | tyrok | Trying Echo... |
| 05:05.10 | tyrok | Getting no audio on the client end, but it's not disconnecting. |
| 05:05.26 | p3nguin | Sounds like a NAT problem. |
| 05:06.20 | tyrok | It is NAT'ed, and I configured it as such. When I look at the debug log, it logs that it's sending the external address instead of the internal. |
| 05:06.38 | p3nguin | That's what we like to call a misconfiguration. |
| 05:09.01 | tyrok | PlayTones() works. |
| 05:09.20 | p3nguin | Sounds like one-way audio to me. That's almost certainly a NAT problem. |
| 05:09.58 | tyrok | Well, that's at least something I can search for. Thanks for the help! |
| 05:14.32 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 05:26.46 | phix | gang!!1 |
| 05:30.14 | *** part/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 05:34.02 | *** join/#asterisk ChannelZ (channelz@burner.com) |
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| 06:14.49 | [TK]D-Fender | tyrok: pastebin a failed attemp with SIP debug enabled |
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| 06:31.12 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 07:28.17 | irroot | anyone here know anything about vibe ?? someone telling me you can put 24 calls on a 1mbs DSL link ... gave them a good price on london bridge suspect i may have sold it |
| 07:28.17 | Nugget | What am I gonna do with 40 subscriptions to Vibe?! |
| 07:32.09 | kaldemar | irroot: seems they even promise 100 concurrent calls on a 2 Mbps link and still support data traffic. |
| 07:32.50 | irroot | kaldemar yeah ill take 10 :P |
| 07:34.54 | kaldemar | they do say it would be on G.729a, but 100 is still just ridiculous. |
| 07:43.43 | ChannelZ | the math works on paper, but in practice... hmm |
| 07:44.16 | irroot | kaldemar it seems to create a tunnel of some sort GRE perhaps and sends similar to IAX packets headerless combined as many in one packet as they can |
| 07:45.20 | irroot | ChannelZ kaldemar in practice its a bugger up on contended service as the packets get shaped and net quality is terrible |
| 07:45.40 | irroot | that was with 4 calls |
| 07:49.40 | kaldemar | yep, the bandwidth requirement difference on 100 calls between trunked and non-trunked IAX for example is huge. without some sort of trunking, it wouldn't work even on paper. |
| 07:49.42 | *** join/#asterisk nix8n82-phone (~AndChat@71-32-137-67.chyn.qwest.net) |
| 07:51.18 | *** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4) |
| 07:56.05 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
| 07:56.51 | _N1x | hi guys , i have one problem , i have downloaded addons , compiled and installed but can't see in asterisk ( show module like ) |
| 07:57.15 | _N1x | i need cdr_mysql addon |
| 07:57.32 | ollii | did you make "make menuselect"? |
| 07:57.37 | ollii | cd asterisk-addons-xx |
| 07:57.44 | _N1x | and when i making make samples , output is Config file configs/cdr_mysql.conf.sample is unchanged |
| 07:57.44 | ollii | $make menuselect |
| 07:57.53 | ollii | check if its activated |
| 07:58.15 | _N1x | sec |
| 07:59.03 | _N1x | ollii: No menuselect user interface found. Install ncurses, |
| 07:59.05 | ollii | you also have to make sure that mysqlclient is installed |
| 07:59.14 | ollii | libncurses-dev on ubuntu |
| 07:59.26 | _N1x | debian |
| 07:59.41 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:59.42 | schmidts | good morning |
| 07:59.58 | *** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de) |
| 08:00.09 | _N1x | ollii: what can i do? |
| 08:00.12 | LiuYan | ~book |
| 08:00.12 | 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 |
| 08:02.44 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 08:02.55 | kaldemar | _N1x: which version of asterisk are you using? |
| 08:03.06 | singler | _N1x: install libncurses5-dev |
| 08:03.37 | _N1x | kaldemar: 1.6.2.9 |
| 08:03.40 | singler | also you will need libmysqlclient?-dev |
| 08:04.15 | *** join/#asterisk pietro (~pietro@88-149-224-186.dynamic.ngi.it) |
| 08:04.47 | _N1x | singler: i have installed both |
| 08:05.04 | singler | run ./configure and then make menuselect |
| 08:05.16 | singler | after configure check if it was successful |
| 08:05.40 | _N1x | singler: in addons folder? |
| 08:05.53 | singler | yes |
| 08:06.19 | ollii | ./configure --with-ncurses=/usr/lib did work for me ... had some problem on ubuntu with recognizing libncurses |
| 08:06.31 | _N1x | singler: http://pastie.org/2845984 |
| 08:07.06 | *** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se) |
| 08:07.38 | _N1x | hm |
| 08:07.38 | singler | hm, never had such problem. are you sure libncurses5-dev is installed? |
| 08:07.41 | _N1x | sec |
| 08:07.50 | _N1x | yes |
| 08:11.56 | LiuYan | hi all, did you encountered asterisk console been frozen/halt (can't input, no output)? |
| 08:18.13 | *** join/#asterisk JohnTeddy (unstable@tor/regular/sid) |
| 08:18.24 | _N1x | singler: nothing happened :( |
| 08:18.37 | *** join/#asterisk bulkorok (d9468199@gateway/web/freenode/ip.217.70.129.153) |
| 08:19.12 | *** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk) |
| 08:22.38 | JohnTeddy | I am from NY, but living in Shanghai. Phones are annoying in Shanghai.. since the Internet sucks a bit. The best method I've found to use the phones.. is for Hong Kong land line to call a Shanghai landline/cell. I do this by using a google voice call back, so in voice.google.com I tell it the American number I want to call, (already in my voice settings is a skype number). Google calls the skype number first, which connections say NY<>Hong Kong, then |
| 08:22.50 | JohnTeddy | I don't know if the ircd char limit chopped that off, if it did let me know. |
| 08:23.46 | JohnTeddy | So what I want to do now.. is setup a call center in Shanghai.. but only with say 3 people. So I want to have nice phones, and perhaps an asterisk box... is there a way I can integrate this 'best known method' for calling Shanghai<>America.. into an asterisk setup? |
| 08:32.59 | plundra | JohnTeddy: Kong, then<END> |
| 08:33.13 | plundra | JohnTeddy: If you're using irssi, try splitlong.pl. |
| 08:37.13 | JohnTeddy | Google calls the skype number first, which connections say NY<>Hong Kong, then Hong Kong calls my Shanghai local phone. The bandwidth between Hong Kong/New York is great.. and the call is usually excellent. |
| 08:38.43 | JohnTeddy | So since the bw between NY<>Shanghai is bad (because of Shanghai), I do NY<>Hong Kong through skype (since that bw connection is great). Skype calls me locally from Hong Kong <> Shanghai, which is also ok. |
| 09:02.29 | *** join/#asterisk Rholk (~Rholk@bny92-1-82-67-178-101.fbx.proxad.net) |
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| 09:55.35 | mandla | tzafrir, can a mobile phone router be integrated with xorcom astribanks?? |
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| 09:56.19 | tzafrir | mandla, what type of mobile phone? wifi or cellular? |
| 09:59.02 | mandla | tzafrir, cellular. |
| 10:06.42 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
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| 10:11.43 | AdvoWork | Hi there, ive got an extension that ive assigned music on hold for, which is definately set, but its still only playing the default MOH. ANy ideas? |
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| 10:28.56 | *** join/#asterisk qakhan (~qakhan@182.185.251.175) |
| 10:29.11 | qakhan | hi |
| 10:29.38 | qakhan | i have a question about callback agent login in queue |
| 10:30.25 | qakhan | i want to check if there is any callback agent login in queue then call should be handle by agent |
| 10:31.01 | qakhan | if there is no callback agent available in queue then call go to some other extension |
| 10:31.16 | qakhan | can anyone help me in this regards? |
| 10:34.44 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:ecb5:f3b0:b617:cadc) |
| 10:40.07 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 10:40.40 | as001 | hi my Asterisk 1.6.2.20 just crashed with following error in messages: rtp.c: RTCP RR transmission error, rtcp halted: Bad file descriptor |
| 10:41.38 | as001 | do you know something more about it ? |
| 10:44.43 | *** join/#asterisk gordonjcp (~gordonjcp@aramaki.gjcp.net) |
| 10:48.13 | qakhan | any help? |
| 10:48.19 | as001 | This seems to be second crash of this week with that error |
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| 10:59.53 | *** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk) |
| 11:08.11 | SeRi | as001, does asterisk crash all together or you are having issues with RTP? |
| 11:18.23 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 11:18.50 | *** join/#asterisk krotos (~d3v1l@host211-3-dynamic.5-87-r.retail.telecomitalia.it) |
| 11:18.54 | krotos | hi :) |
| 11:20.48 | krotos | i've got my asterisk 1.8 (svn irrot branch) that stay at 45% of cpu-usage when there are two concurrent call? there is some way to understand why? |
| 11:24.12 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
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| 11:37.21 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 11:50.58 | schmidts | krotos what kind of calls? |
| 11:55.42 | as001 | asterisk crash and I am not able to connect to cli |
| 11:56.18 | as001 | i get unable to connect check if process exist.. etc.. |
| 12:00.37 | kaldemar | as001: is it running? |
| 12:02.29 | as001 | after I restart it it is running |
| 12:03.14 | as001 | before restart I am unable to connect and I guess it is not running |
| 12:03.47 | *** join/#asterisk k3asd` (~k3asd@host139-84-dynamic.35-79-r.retail.telecomitalia.it) |
| 12:03.47 | bulkorok | check with ps ax |
| 12:04.09 | as001 | i can't now I restarted it now it has process id for sure |
| 12:04.14 | kaldemar | as001: if a program crashes, it rarely is running after the crash until it is started again. |
| 12:04.46 | as001 | Next time when it crash I can try to check if process exists or not |
| 12:05.25 | kaldemar | don't bother. if it really crashes and nothing starts it again, it will not be running. |
| 12:06.05 | kaldemar | you might want to consider using safe_asterisk if you want it to start again automatically. |
| 12:06.37 | as001 | I would like to avoid or to prevent that crash and that error message somehow |
| 12:07.25 | as001 | Of course I can easily restart when it happens but 50 people is on my neck when it happends because their calls are stopped... |
| 12:08.37 | schmidts | does anyone of you have a CFCA account? i need a special report from there |
| 12:18.16 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 12:21.07 | k3asd` | hi |
| 12:25.21 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 12:52.06 | *** join/#asterisk _N1x (~n1x@mail.orient-logic.com) |
| 12:52.15 | _N1x | hi guys , i have problem with sipp |
| 12:52.52 | _N1x | when i making concurrent calls , im reciving this message (unexpected bye recived from xxxxx ) |
| 12:53.18 | _N1x | my cpu isn't loaded but i have not more than 70 concurrent calls |
| 12:53.44 | _N1x | but i have cps calls |
| 12:54.35 | schmidts | _N1x maybe a call limit on asterisk? |
| 12:55.18 | _N1x | schmidts: i have not specified parrametrs , only ulimit |
| 12:55.21 | _N1x | for open many files |
| 12:55.54 | _N1x | and i have edited limits.conf file also... |
| 12:56.20 | *** join/#asterisk Akiraa (~Akiraa2@unaffiliated/akiraa) |
| 12:57.04 | Akiraa | has someone installed asterisk on a router with openwrt or dd-wrt? I was wondering if this: http://wiki.openwrt.org/toh/tp-link/tl-wr1043nd?s=tplink could support 5 extensions and 2 trunks |
| 12:58.30 | kaldemar | Akiraa: depends on what you do with it. |
| 12:58.54 | Akiraa | kaldemar: needs to work without major pains |
| 12:58.58 | schmidts | _N1x where do you see this unexpected bye message, on the sipp side or in asterisk? and what do you see in your asterisk cli |
| 13:00.01 | kaldemar | Akiraa: what needs to work? asterisk will run on it, but depends on how you use it. for calls without transcoding or protocol switching it will most likely be sufficient. |
| 13:00.02 | lcat | hi guys, someone are using FastAGI + Haproxy ? |
| 13:00.29 | Akiraa | kaldemar: protocol switching, as in translating from sip to iax2? |
| 13:00.56 | kaldemar | Akiraa: for example, yes. |
| 13:01.21 | lcat | I get timed out after MAX_AGI_CONNECT all the time |
| 13:03.01 | *** part/#asterisk lcat (~lcat@187.45.254.21) |
| 13:03.12 | _N1x | schmidts: on the sipp yes |
| 13:03.42 | _N1x | schmidts: nothing in asterisk , but when im increasing call limits with my hands i see , maximum transmiss is excided |
| 13:03.53 | *** join/#asterisk lcat (~lcat@187.45.254.21) |
| 13:04.11 | Akiraa | does asterisk require root access to be installed? |
| 13:04.31 | _N1x | schmidts: i noticed when im using -l option on the sipp my concurrent calls isn't more than 70 |
| 13:04.54 | schmidts | _N1x could you show me your sipp command |
| 13:05.25 | *** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
| 13:05.46 | _N1x | schmidts: yes sec , i have used many different commands |
| 13:06.18 | _N1x | schmidts: sipp -sn uac 192.168.8.33 -s 2000 -d 10000 -l 1000 |
| 13:08.03 | _N1x | schmidts: and when i'm using this command ( sipp -sn uac 192.168.8.33 -s 2000 -r 500 -d 1500 |
| 13:08.22 | _N1x | i have 500 cps and 750 conncurent calls simultaneously |
| 13:10.11 | *** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net) |
| 13:10.53 | schmidts | _N1x the -l param is only the limit overall |
| 13:11.46 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:12.00 | _N1x | schmidts: ye |
| 13:12.01 | _N1x | s |
| 13:15.59 | schmidts | i will test this |
| 13:16.06 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 13:16.27 | *** join/#asterisk nighty^ (~nighty@TOROON12-1279662182.sdsl.bell.ca) |
| 13:16.55 | schmidts | it works for me |
| 13:17.50 | schmidts | i only have to use the -i param cause sipp couldt alloc an audio port if i dont use this, but it works for me |
| 13:17.59 | schmidts | how does your extension 2000 look like? |
| 13:19.13 | *** join/#asterisk mjordan (~mjordan@nat/digium/x-magbeooyjqcldqya) |
| 13:19.26 | ollii | is there a real irc chan for hylafax? i ran into a issue where i lookup a name <-> facsimile number and hylafax sends me all of its device if a name has a "*" in it |
| 13:20.55 | ollii | CALLERID(name) = foo * dude 1 results in: foo ttyIAX1 ttyIAX0 .. dude 1 |
| 13:21.12 | ollii | CALLERID(name) = foo =* dude 1 results in foo =* dude 1 |
| 13:23.41 | schmidts | olliii strange ;) neer seen this problem |
| 13:23.54 | schmidts | is this for incoming our outgoing faxes? |
| 13:24.58 | ollii | incoming |
| 13:27.00 | [TK]D-Fender | ollii, it would appear to want the * with spaces around it |
| 13:27.34 | ollii | http://pastebin.com/3NsirWin |
| 13:28.01 | ollii | ah |
| 13:28.06 | ollii | its /var/spool/hylafax folder |
| 13:28.11 | ollii | from where the script is called |
| 13:28.19 | ollii | okay so i should transform it in my lookup |
| 13:29.28 | SeRi | morning all |
| 13:32.20 | _N1x | schmidts: fixed maybe , i remove hangup at end of my extenison |
| 13:32.25 | _N1x | and set musiconhold |
| 13:35.23 | ollii | [TK]D-Fender: thats the problem...http://pastebin.com/87UDyN0b |
| 13:35.49 | ollii | maybe i masquerade "*" with an "\" |
| 13:36.36 | ollii | echo \* => * |
| 13:40.07 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 13:48.06 | *** join/#asterisk cerberus_za (~coert@8ta-151-38-115.telkomadsl.co.za) |
| 13:50.23 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:50.35 | ollii | Nov 11 13:02:42 hostname FaxGetty[4598]: RECV FAX: bin/faxrcvd "recvq/fax000002860.tif" "ttyIAX0" "000010468" "" "05066808249" "Foo * Dude GmbH" "<NONE>" "20" |
| 13:50.49 | ollii | FaxGetty sends it correctly to faxrcvd script |
| 13:51.07 | ollii | so the problem has to be somewhere in the generation of the subject |
| 14:04.44 | tyrok | p3nguin: Actually, it looks like that problem was not due to NAT but the firewall. |
| 14:06.21 | *** join/#asterisk becca_r (~IceChat77@72.165.148.230) |
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| 14:08.05 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 14:11.23 | tyrok | RTP was not being allowed in by iptables. Now that it is, it no longer drops the call when I run Festival(). I get no audio yet, but at least it's a different problem. |
| 14:14.39 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 14:16.30 | becca_r | morning all |
| 14:17.30 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-mgvmabahgdpvftej) |
| 14:17.40 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:17.40 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:20.52 | *** join/#asterisk tehrcon (~tercon@mail.imprinters.com) |
| 14:23.25 | *** join/#asterisk dijib (~nobodysho@bas10-kitchener06-1279682439.dsl.bell.ca) |
| 14:23.29 | dijib | Everyone is busy/congested at this time (1:0/1/0) |
| 14:23.44 | dijib | anybody have any idea why today i wake up and im getting this on outbound calls |
| 14:23.52 | dijib | there was a power failer last night |
| 14:24.15 | dijib | but the dialplan is intact, the SIP registry is online |
| 14:24.20 | dijib | i dont get it |
| 14:24.30 | dijib | oh also i can recieve inbound calls |
| 14:24.58 | kaldemar | no idea at all unless you provide more debug output. |
| 14:25.37 | irroot | could help if you reset the NT im assuming its a T1/E1 span |
| 14:26.57 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 14:27.28 | dijib | i enabled sip debug... but then the calls started going through... without any real change |
| 14:28.42 | Naikrovek | becca_r: morning |
| 14:28.46 | Naikrovek | dijib: probably wasn't you, then. |
| 14:28.52 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 14:29.03 | Naikrovek | my sip provider chokes all the flippin time |
| 14:29.34 | dijib | very weird |
| 14:29.38 | dijib | who is your ITSP? |
| 14:29.55 | Naikrovek | we buy from someone that resells level3 sip trunks |
| 14:30.09 | Naikrovek | but they, and level3 consistently screw things up |
| 14:30.50 | Naikrovek | g729 calls work one day, but not the next. international dialing works one day but not the next. incoming works one day but not the next |
| 14:30.53 | dijib | im with itsp, and i have yet to this day seen an outage like this |
| 14:32.10 | [TK]D-Fender | dijib, That message offers no details, registration has nothing to do with authing calls or validating your peer |
| 14:32.45 | [TK]D-Fender | dijib, raw statement about getting inbound also proves little by itself. Time to look at the actual call. |
| 14:37.06 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:37.30 | dijib | since the calls are now going through |
| 14:37.40 | dijib | the issue was not on my end |
| 14:40.32 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
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| 14:56.25 | *** part/#asterisk tyrok (~tyrok@c-68-41-70-233.hsd1.mi.comcast.net) |
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| 15:15.42 | eppigy | mornin |
| 15:18.44 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 15:25.37 | p3nguin | [tk]d-fender: Actually, in this case, registration does have something to do with calling. VoIP.ms (which he is using), as well as other ITSPs, require the device to be registered before it is allowed to make calls. I'm guessing it is because they configured SER to behave this way (since that is not a feature of Asterisk). |
| 15:26.14 | *** part/#asterisk irroot (~irroot@pbx.distrotech.co.za) |
| 15:27.54 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:27.59 | [TK]D-Fender | p3nguin, Still proves nothing since he wasn't showing any real debug though. |
| 15:29.01 | *** join/#asterisk scubes13 (~scubes13@rrcs-96-10-101-236.se.biz.rr.com) |
| 15:29.18 | p3nguin | Oh I know. |
| 15:29.39 | p3nguin | I wasn't trying to say it proved anything. Just saying in this case registration does matter. |
| 15:33.25 | [TK]D-Fender | p3nguin, I suppose... would be at least a little more valid if he explicitly mentioned using them with this tidbit being of any common knowledge... but that'd be getting ahead of myself :) |
| 15:35.13 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 15:35.25 | dijib | either way it resolved itself. |
| 15:44.59 | dijib | foot on snow outside today |
| 15:45.33 | Qwell | foot on snow? |
| 15:45.38 | dijib | of |
| 15:45.39 | Qwell | that sounds rough. |
| 15:45.40 | dijib | sorry |
| 15:45.48 | Qwell | wake up, discover severed foot laying on snow |
| 15:45.54 | dijib | hahaha |
| 15:47.19 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 15:50.19 | [TK]D-Fender | dijib, Just because the symptom went away doesn't mean the problem has. |
| 15:51.20 | dijib | since nothing changed with my setup, i believe the issue to be a voip.ms issue. not mine. |
| 15:51.41 | dijib | in that i believe they have resolved the issue they were having, as i have confidence in their abilities |
| 15:52.14 | [TK]D-Fender | Qwell, http://www.youtube.com/watch?v=8qWFhDvURLg |
| 15:53.29 | [TK]D-Fender | dijib, well you've mastered the use of faith over debugging then... |
| 15:54.01 | dijib | it was funny as soon as i did a 'sip set debug on' it started working |
| 15:55.40 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 15:57.37 | r0m|u | waz up guys |
| 15:57.42 | Naikrovek | either way, there's no longer a problem to diagnose |
| 15:57.43 | *** join/#asterisk irroot (~irroot@41.49.200.129) |
| 15:59.05 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 15:59.29 | r0m|u | [TK]D-Fender, Its all fixed. It turned out I was right the damn device has bugs! It send a "REGISTER" instead of an "INVITE" when registering is truned off :) ether way I found how it was sending the cid and it was in the cid=name field. I am able to catch the number that way. p3nguin helped a lot! |
| 16:00.13 | [TK]D-Fender | r0m|u, \o/ |
| 16:00.27 | r0m|u | ~motorboat |
| 16:00.28 | infobot | <r0m|u> its all clear now. like boobs slaping you in the face! |
| 16:00.34 | r0m|u | :) |
| 16:01.05 | *** part/#asterisk iEatChildren (~iEatChild@204.26.87.226) |
| 16:01.07 | r0m|u | [TK]D-Fender, thanks for the help. |
| 16:01.17 | r0m|u | dijib, you got your issue resolved? |
| 16:01.52 | dijib | yessum |
| 16:02.27 | r0m|u | cool |
| 16:07.27 | r0m|u | dijib, just curious what was the issue? your ata? |
| 16:08.49 | [TK]D-Fender | r0m|u, can't prove, never really looked :) |
| 16:09.12 | [TK]D-Fender | And it "magically" fixed itself (also the way they break. Call it "convenience") |
| 16:09.50 | *** join/#asterisk Foxi352 (~Foxi352@ppp-104-188.adsl.restena.lu) |
| 16:10.11 | r0m|u | ....?.... |
| 16:11.33 | dijib | r0m|u, no clue |
| 16:11.40 | dijib | im heading out now for 4 hours.. .ill be back |
| 16:11.51 | r0m|u | A wise man once said "Issues are proved with facts and not "Magic" or "Faith"! |
| 16:11.57 | irroot | i like people who pay to get a job done but do it them self so they can learn .... |
| 16:11.58 | r0m|u | cya |
| 16:12.28 | r0m|u | irroot, school? |
| 16:13.00 | r0m|u | Those can be the most complicated customers some time. |
| 16:13.43 | *** part/#asterisk sekil (~sekil@78.24.104.73) |
| 16:14.13 | Foxi352 | Someone ever managed to get the "Cisco Mobile" iphone app running on asterisk server ? |
| 16:14.26 | irroot | lol ironnically it was for a school catholic girls boarding school :P the network admin there is rather jacked and went IP wanted to switch over to our distro and fix up problems the install has been a mess from start and the ones doing it are out of bussiness i like this kind of customer |
| 16:15.59 | p3nguin | ~faith |
| 16:16.00 | infobot | Telephony is not faith-based. Look. Always look, and then show. |
| 16:16.20 | irroot | ~beer p3nguin |
| 16:16.20 | infobot | ACTION deftly decants a fine Jever for p3nguin |
| 16:16.27 | Naikrovek | why can't you just say it yourself? i don't understand these shortcuts |
| 16:17.39 | wdoekes2 | *sings* jag känner an booot! hon heter Anna, Anna heter hooon |
| 16:17.52 | _Corey_ | irroot: My favorites are the customers who do their own research and tell you how to do something rather than ask how you would do it... :) |
| 16:18.41 | Foxi352 | And if you do it like they say, and it does not work, you are the one that does not know what he does :-) |
| 16:19.21 | _Corey_ | They just never work out well because we end up spending hours explaining why they're wrong and nobody likes to hear that they're wrong... ;) |
| 16:19.27 | irroot | _Corey_ as long as they not the ones who google land up on voip-info and come and tell you what is wrong with the way you do it |
| 16:21.16 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 16:21.18 | _Corey_ | irroot: Usually it's the ones who expect to use Switchvox or stock FreePBX to run a 150 agent call center, bought it from another vendor who couldn't make it work, and then expect you to make it work |
| 16:21.35 | Qwell | _Corey_: ugh |
| 16:21.57 | r0m|u | p3nguin, I didnt know you embedded it to the bot. nice lol |
| 16:22.12 | r0m|u | irroot, Hey as long as they pay is all good :) |
| 16:23.11 | irroot | _Corey_ i format reinstall or walk away ... it ends badly when you bail them out ... also but its free why must we pay ... |
| 16:23.17 | _Corey_ | Qwell: eh, at least we get paid to clean up messes like that. I just spent the last 25 minutes on the phone with a carrier explaining what RDNIS is... |
| 16:23.43 | irroot | notes for the record we do not sell software but offer onsite support config / ... service |
| 16:23.54 | _Corey_ | I may have to start drinking in the morning if this trend continues |
| 16:24.16 | Qwell | _Corey_: sounds like it's a win-win situation |
| 16:24.26 | irroot | _Corey_ did 2 beers before 13h |
| 16:24.26 | Qwell | You'd be getting paid to drink. |
| 16:25.08 | r0m|u | I want to get paid to dinrk |
| 16:26.28 | *** join/#asterisk billmania (~bill@38.98.130.98) |
| 16:26.31 | irroot | but i love users ... the one who called telling us there system was off we got it up logged in to find the power button was pressed as per acpid logs yet no one was near it or was in the server room .... ghosts ? |
| 16:26.53 | irroot | that is why we drink |
| 16:27.02 | r0m|u | no ghost's.... monkeys! |
| 16:27.11 | Qwell | irroot: I drink to forget |
| 16:27.28 | *** join/#asterisk mmcji (~mmcji@65.172.54.254) |
| 16:28.01 | irroot | mmm could be baboons there are baboons in those parts but not in office areas was a sea lion that made a home in one so its possible |
| 16:28.42 | r0m|u | lmao |
| 16:28.46 | billmania | What causes sig_pri.c in asterisk 1.8.4.2 to log this: Received NOTIFY on unconfigured channel 255/255 |
| 16:29.09 | billmania | I see that it's the PRI_EVENT_NOTIFY event but I don't understand the source of that "event". |
| 16:29.34 | irroot | billmania all ones in d channel has been my guess but would like to see |
| 16:30.05 | irroot | possibly a telco test or NT fault are second guesses |
| 16:30.32 | billmania | Where does one look to learn more about the cause? I've been seeing these in my asterisk verbose log for several weeks. |
| 16:36.14 | *** join/#asterisk brdude (~brdude@12.155.183.30) |
| 16:37.01 | *** join/#asterisk endemic (~endemic@2001:49f0:400e::5) |
| 16:40.07 | r0m|u | billmania, http://www.google.com/search?sourceid=chrome&ie=UTF-8&q=unconfigured+channel+255%2F255 |
| 16:40.28 | billmania | r0m|u: Looking now. Thanks. |
| 16:41.02 | r0m|u | n/p there seems to be a lot stuff about that on the web. |
| 16:41.05 | r0m|u | ~google |
| 16:41.05 | infobot | i heard google is http://google.com |
| 16:42.04 | r0m|u | ~Friday |
| 16:42.05 | infobot | i guess friday is !!!!!!!! FRIDAY !!!!!!! |
| 16:42.26 | r0m|u | hells yea bot! its fucking friday! |
| 16:43.51 | irroot | can some one add ~google http://justfuckingoogleit.com |
| 16:44.07 | [TK]D-Fender | ~jfgi |
| 16:44.08 | infobot | http://www.google.com/search?q=jfgi |
| 16:44.15 | [TK]D-Fender | OLD |
| 16:44.20 | irroot | www.justfuckinggoogleit.com |
| 16:44.29 | r0m|u | rofl! |
| 16:44.44 | billmania | r0m|u: While there is indeed a lot of mention. I've already seen a lot of them but discounted most of them because they seem to be several years old. |
| 16:46.57 | *** join/#asterisk jrose_atDigium (~jon@nat/digium/x-shbvoirmzbbwxkai) |
| 16:47.12 | irroot | billmania it happens frequent |
| 16:47.14 | irroot | ? |
| 16:48.07 | billmania | irroot: Are you asking if this PRI_EVENT_NOTIFY occurs frequently with my system or telling me that this is a frequent PRI condition? |
| 16:48.17 | irroot | yes |
| 16:49.03 | *** join/#asterisk blebleble (~blebleble@s7.n194.n222.n216.static.myhostcenter.com) |
| 16:49.18 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 16:49.54 | blebleble | How can I see how many times a inbound route has been dialed? I looked throughout he master.csv however it doesn't seem to actually track it. I did some test calls and it only lists the caller id of the caller and then the end destination channel not the number it self |
| 16:51.47 | r0m|u | blebleble, The Extension is the destination. |
| 16:52.04 | r0m|u | It all depends how you are logging with cdr |
| 16:52.51 | irroot | blebleble may i recomend using userfield and accountcode to simplify this |
| 16:53.27 | r0m|u | in mine its Date-Tim ---- CID Name ---- CID Num ---- Extension (Destination) |
| 16:53.53 | blebleble | http://pastebin.ca/2093837 - basically 61411111111 was the inbound calling number that dialed 6141111112 yet the 1111112 does not show only the 9558 end destination extension |
| 16:56.57 | *** part/#asterisk [TK]D-Fender (~aoulton@216.191.106.162) |
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| 17:06.49 | blebleble | irroot: can you describe what you've done to make that work / happen? or point me to some documentation |
| 17:10.17 | irroot | core show function CDR blebleble type that in the "asterisk -r" console |
| 17:11.38 | irroot | if you want to track a record set the userfield to what you want to track latter ie the number you referring too if you want to have someone account for the call either by billing or report or some reason search for it set the account code |
| 17:21.11 | r0m|u | quiet Friday :) |
| 17:23.23 | r0m|u | strange nick ^^ |
| 17:28.25 | *** join/#asterisk RobH (~RobH@wikimedia/RobH) |
| 17:30.38 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 17:32.37 | as001 | Hello I failed to register to my sip trunk with asterisk 1.8.7.1. I have forbiden bad password in sip show debug... In one tutorial I found that Asterisk 1.6 can not send @ (et) in sip header for registration and my provider gave me username phonenumber@theirhostname so I must send @.. Asterisk 1.4 according to tutorial can send @ so I need to downgrade to 1.4... Do you know something about that ? |
| 17:33.33 | as001 | I am sure I am sending to them what they said is my authusername and password... |
| 17:34.08 | as001 | register => 123xxxxx@their.host.name:pass@10.0.0.2 |
| 17:34.43 | as001 | register => 123xxxx@their.host.name : password @ 10.0.0.2 without spaces |
| 17:35.37 | as001 | tutorial said @ in authname is problem for asterisk 1.6 is that true ? I can not believe i ned to install old 1.4.x.... |
| 17:37.34 | *** join/#asterisk garymc (~chatzilla@81.138.225.164) |
| 17:38.14 | irroot | as001 you cant ask them to reset password without @ |
| 17:38.18 | as001 | here is pastebin from sip debug http://pastebin.com/ea391R0y |
| 17:38.48 | as001 | i plan to do that but they are very stubborn monopolic company I doubt they will listen.. |
| 17:39.21 | as001 | they will rather say "use cisco" or something like that.. |
| 17:44.33 | *** part/#asterisk as001 (~uros@82.117.198.142) |
| 17:46.27 | *** join/#asterisk as001 (~uros@82.117.198.142) |
| 17:46.49 | as001 | so what do you think could that thing with @ be a problem in registration ? |
| 17:47.02 | as001 | is it known issue for Asterisk> 1.4 ? |
| 17:48.15 | irroot | have had this problem yes and got the password changed as001 tell them you think the password was leaked and want a new one :P |
| 17:49.10 | *** join/#asterisk devmikey (~irc@unaffiliated/devmikey) |
| 17:50.15 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 17:52.08 | as001 | Ok I will try but I see myself downgrading to 1.4...:-( |
| 17:52.14 | as001 | thanks for help |
| 17:52.44 | irroot | as001 did you find a fault on the bug tracker ? |
| 17:53.01 | as001 | i didnt search there |
| 17:53.54 | irroot | hang arround as001 someone more familiar may be arroun |
| 17:55.48 | as001 | I believe you and that tutorial I will try with 1.4 to see if it can register and if it can i know what is the problem... |
| 17:56.17 | as001 | i must log off... Bye |
| 17:56.31 | *** part/#asterisk as001 (~uros@82.117.198.142) |
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| 19:13.04 | r0m|u | yawns |
| 19:31.06 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 19:33.07 | *** join/#asterisk TelephonyJr (~chatzilla@c-98-234-122-194.hsd1.ca.comcast.net) |
| 19:33.17 | TelephonyJr | hi everyone |
| 19:33.25 | TelephonyJr | I am having some trouble figuring something out |
| 19:33.30 | TelephonyJr | I am creating an automated dilaer |
| 19:33.31 | TelephonyJr | dialer |
| 19:33.37 | TelephonyJr | where I have two sales reps |
| 19:33.43 | TelephonyJr | and 8 outbound lines that are making calls to direct phone numbers |
| 19:33.55 | TelephonyJr | upon reaching the prospect, the call will be instantaneously switched to one of the available sales reps |
| 19:34.05 | TelephonyJr | if no sales reps are available the calling will stop |
| 19:34.14 | TelephonyJr | I was reading the California telemarketing regulations |
| 19:34.16 | TelephonyJr | and found out that |
| 19:34.23 | TelephonyJr | not more than 3% of calls can be abandoned within a 30 day period |
| 19:34.32 | TelephonyJr | does anyone know of a creative way to be compliant with this |
| 19:34.59 | TelephonyJr | i am concerned that since 8 lines are calling at once, it is very likely that more than 2 calls will be connected with simultaneously |
| 19:35.05 | TelephonyJr | causing an abandoned call |
| 19:35.19 | TelephonyJr | does anyone know what i can do in this situation? |
| 19:38.35 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 19:40.19 | [TK]D-Fender | </monologue> |
| 19:40.53 | r0m|u | don't like telemarketers. |
| 19:40.57 | [TK]D-Fender | TelephonyJr, Only call out when you have available agents. |
| 19:45.09 | *** join/#asterisk singler (~singler@84.15.186.239) |
| 19:45.44 | [TK]D-Fender | TelephonyJr, 8 lines is therefor a waste |
| 19:47.48 | r0m|u | lol |
| 19:48.19 | TelephonyJr | Fender: but that reduces the ratio leverage |
| 19:48.22 | *** join/#asterisk joshaidan_ (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 19:48.31 | TelephonyJr | Fender: if it's 1:1 there is no real benefit |
| 19:49.28 | [TK]D-Fender | TelephonyJr, If you call out 3 line for 2 agetns and every line gets answered then you have 33% abandoned and a dead legally. |
| 19:49.32 | *** join/#asterisk Ast001 (~uros@cable-89-216-178-211.dynamic.sbb.rs) |
| 19:50.02 | [TK]D-Fender | Simple math |
| 19:50.04 | *** join/#asterisk garymc (~chatzilla@host31-53-157-167.range31-53.btcentralplus.com) |
| 19:50.06 | TelephonyJr | haha that's true |
| 19:50.19 | TelephonyJr | Fender: Do you know the statistics or where I could find them, for when there are 5 agents available I can have X lines calling, when only 4 are available, Y, etc? |
| 19:50.43 | [TK]D-Fender | Agents available = max calls out allowed. |
| 19:50.44 | [TK]D-Fender | X=y |
| 19:50.51 | [TK]D-Fender | Not much of a formula. |
| 19:50.56 | TelephonyJr | then what is the purpose of a predictive dialer? |
| 19:51.02 | Ast001 | just for record if anyone else have same problem in some banana state like mine is tutorial was right asterisk 1.4 registers without problem with register => "number@host": pass@ip while asterisk 1.8 and 1.6 can not send @ in number @ host |
| 19:51.03 | TelephonyJr | i thought that the purpose was to allow for leverage |
| 19:51.16 | TelephonyJr | if it is 1:1 ratio, there really is no add'l throughput |
| 19:51.27 | [TK]D-Fender | TelephonyJr, The term "predictive dialer" has no implication on legor or practical capacity |
| 19:51.36 | *** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com) |
| 19:51.37 | Ast001 | so I will be proud user of 1.4 :( thanks to telco |
| 19:51.41 | [TK]D-Fender | TelephonyJr, You are the one creating this false association. |
| 19:51.52 | TelephonyJr | I see |
| 19:51.54 | TelephonyJr | you are correct |
| 19:52.02 | TelephonyJr | i thought that the usage would be legal |
| 19:52.05 | TelephonyJr | automatically |
| 19:52.15 | TelephonyJr | but it looks like in CA ,at least, the benefits can not be realized? |
| 19:52.17 | TelephonyJr | is that fair to say? |
| 19:52.21 | [TK]D-Fender | A dialer dials. Predictive just means it's smarter. Now if you want to try to do something stupid with it, then that's your prerogative. |
| 19:52.29 | TelephonyJr | i understand |
| 19:52.39 | TelephonyJr | Thanks, Fender |
| 19:52.42 | TelephonyJr | for the clarifiction |
| 19:52.52 | TelephonyJr | i was wondering, other than the leverage that I thought would be of huge value |
| 19:52.54 | [TK]D-Fender | The benefits of a 100 line/2 agent system sure won't be ... in CA |
| 19:53.00 | TelephonyJr | what add'l benefits can the predictive model provide? |
| 19:53.12 | [TK]D-Fender | Stop thinking. You're doing it wrong :p |
| 19:53.24 | r0m|u | lol |
| 19:53.31 | TelephonyJr | Fender: can you help me figure out how to do it correctly? |
| 19:53.45 | [TK]D-Fender | TelephonyJr, "Not at all" :) |
| 19:53.51 | [TK]D-Fender | "Just Say NO" |
| 19:54.00 | TelephonyJr | thanks |
| 19:54.21 | [TK]D-Fender | TelephonyJr, I did suggest the only statistically sound means with is 1:1 |
| 19:54.25 | [TK]D-Fender | which* |
| 19:54.55 | [TK]D-Fender | you could add as you grow agents, but that 3% is a fixed border. |
| 19:55.09 | [TK]D-Fender | 15 lines for 10 agents = bad. I'm sure you'll catch on... |
| 19:55.42 | [TK]D-Fender | Now a dialer just does the dial for you.... so the agent isn't on a paper list. That is still at lease practical in terms of call-setup efficiency. |
| 19:55.52 | [TK]D-Fender | logging, etc |
| 19:57.31 | TelephonyJr | understood |
| 19:58.35 | TelephonyJr | fender: is there a platform that automates the adjusting to match the regulation so that for the two sales reps I could do a 3 lines calling out when there are 2 agents available, and can self-adjust the ratios properly as the campaign progresses to ensure compliance? |
| 19:59.12 | *** join/#asterisk TimeRider (steve@027a41f0.bb.sky.com) |
| 20:03.58 | [TK]D-Fender | Doable (in a longer term), and this would take some minimal counters & math. |
| 20:04.07 | [TK]D-Fender | And * is fine as a platform. |
| 20:08.41 | TelephonyJr | ok fender - thanks! |
| 20:09.24 | *** join/#asterisk Dibbler (~Dibbler@87-194-103-72.bethere.co.uk) |
| 20:10.55 | *** part/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 20:19.18 | *** join/#asterisk Turingi (~devon@unaffiliated/devon-hillard/x-7250961) |
| 20:19.27 | Turingi | who here can whistle DTMF tones? |
| 20:19.41 | r0m|u | my grandma |
| 20:19.49 | [TK]D-Fender | I used to be able to whistle up a 300 baud carrier... |
| 20:19.59 | [TK]D-Fender | Ahh the good 'ole days |
| 20:20.22 | singler | :) |
| 20:20.30 | r0m|u | I wasent around for the "good" days :( |
| 20:20.53 | singler | too |
| 20:22.08 | Turingi | [TK]D-Fender: you are this guy: http://blogs.msdn.com/b/oldnewthing/archive/2011/11/11/10235970.aspx ? |
| 20:24.00 | r0m|u | lol |
| 20:27.36 | [TK]D-Fender | :D |
| 20:27.47 | n3hxs | IMHO, the good old days were only good because you can't remember the "bad". |
| 20:32.52 | carrar | Oh like you can't whistle out SIP packet headers! |
| 20:33.36 | *** join/#asterisk netman (netman@7.227.76.188.dynamic.jazztel.es) |
| 20:34.48 | *** join/#asterisk brotherSand (~da5id@c-98-228-227-159.hsd1.il.comcast.net) |
| 20:35.07 | brotherSand | Hello gang |
| 20:35.41 | brotherSand | I have some questions regarding asterisk. Should I address these questions to anyone in particular or just shout them out? |
| 20:35.51 | Qwell | ~ask |
| 20:35.51 | 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. |
| 20:36.08 | singler | brotherSand: whistle 300 baud carrier and then talk :) |
| 20:36.20 | singler | j/k |
| 20:36.26 | brotherSand | Hehehe |
| 20:37.28 | brotherSand | okay, first off: How does one set up multiple phone numbers to go to an asterisk box? My understanding is fuzzy here. I thought we would have to set up a couple numbers and have a set of extensions. |
| 20:37.55 | brotherSand | Can we actually have a distinct number for each voip phone in the office? |
| 20:38.01 | Qwell | sure |
| 20:38.10 | Qwell | ~book |
| 20:38.11 | 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 |
| 20:39.05 | [TK]D-Fender | brotherSand, Yuo can have analog lines, digital lines, SIP VoIP providers, GSM via BlueTooth, and more all on the same box.... and that's jsu for PSTN connectivity |
| 20:39.15 | [TK]D-Fender | just* |
| 20:39.44 | [TK]D-Fender | brotherSand, Setup up whatever you want to have. process calls however you like. Welcome to * |
| 20:39.59 | r0m|u | brotherSand, Thats an awesome book. Read it you will learn a lot. |
| 20:41.04 | brotherSand | Thank you. I will read it. But just to get the short version, we get a voip provider and they run some form of connection to us. We can then run multiple phone numbers down that connection? |
| 20:42.48 | [TK]D-Fender | brotherSand, Those are the kinds of services most offer, so yes |
| 20:43.21 | brotherSand | Ah, thanks. |
| 20:43.30 | [TK]D-Fender | brotherSand, You want to have a "main" number and pump that through menu's etc, sure. Have direct #'s for employee's desks? Sure. Whatever you want |
| 20:43.49 | [TK]D-Fender | brotherSand, Call comes in. Dialplan lokos at it. Goes and does whatever you tell it to do. |
| 20:44.27 | brotherSand | You see, I got this in an email: "They are using Cisco Call Manager, which is a class above anything we'll be able to do with open source tools. I have no idea if what we're putting in can support individually assigned DID's." Now I have a feeling he is completely wrong, but I need to be able to back up my rebuttal. |
| 20:45.02 | Qwell | brotherSand: they are stupid. |
| 20:45.10 | Qwell | and willing to spend tens of thousands of dollars on garbage |
| 20:45.52 | carrar | but it's cisco garbage :) |
| 20:46.18 | brotherSand | Yeah well I'm working with some Church-of-Microsoft guys here and they don't seem to believe anything they don't spend tens of thousands of dollars on can actually work. |
| 20:46.34 | carrar | hahah |
| 20:46.45 | *** part/#asterisk billmania (~bill@38.98.130.98) |
| 20:46.59 | Qwell | brotherSand: ask Penn State about the tens of thousands of endpoints they moved to Asterisk, *from* Cisco. |
| 20:47.00 | carrar | that has to suck |
| 20:47.22 | Qwell | (I'm sure they'd love different press atm :p) |
| 20:47.30 | carrar | haha |
| 20:47.45 | carrar | I was trying to come up with a smrt remark for that |
| 20:47.48 | brotherSand | yeah, it has drawbacks. |
| 20:48.03 | *** part/#asterisk sorressean (~tyler@tds-solutions.net) |
| 20:48.37 | carrar | how many is tens of thousands |
| 20:48.45 | Qwell | tens of thousands |
| 20:48.50 | carrar | 10,000? |
| 20:48.56 | carrar | or 11,000 |
| 20:48.59 | Qwell | 20+. I don't actually know. |
| 20:49.08 | Qwell | I just know that it's multiple tens |
| 20:49.22 | Qwell | I think it was something like 16,000 in 2007 |
| 20:49.24 | carrar | love to see that setup |
| 20:49.41 | Qwell | I talked to their IT manager at astricon. |
| 20:50.45 | carrar | they running opensips or kamilio to reg the phones? |
| 20:51.49 | Qwell | no idea |
| 20:51.57 | carrar | What did you talk about? |
| 20:52.07 | carrar | weather is number one most talked about item |
| 20:52.12 | Qwell | how cold is was in denver. |
| 20:52.14 | Qwell | :P |
| 20:52.20 | carrar | heh |
| 20:52.44 | Qwell | it was brief smoker chat. we talked about the setup a bit, but not in any detail |
| 20:53.02 | Qwell | I just know that she affirmed "tens of thousands". |
| 20:53.12 | carrar | nice |
| 20:53.23 | carrar | thats awesome |
| 20:53.28 | Qwell | also that they had no f'ing idea what they were doing |
| 20:53.32 | Qwell | but they were doing it anyways |
| 20:53.36 | carrar | haha |
| 20:54.00 | Qwell | they actually have a big team. probably a dozen or so doing asterisk |
| 20:54.37 | *** join/#asterisk happylife (~happylife@212.92.145.7) |
| 20:54.42 | carrar | I imagen most univserities have a a pretty smart unix team |
| 20:56.42 | carrar | When I knew people in the UW networking/unix dept here in WA, they were all pretty smart |
| 20:57.28 | carrar | hence pine & uw-imap |
| 20:58.07 | carrar | alpine nowdays |
| 21:01.36 | _Corey_ | Qwell: Did you meet the PSU guys in the room where we did our presentations? One of my customers ran into them there that morning... |
| 21:01.52 | Qwell | _Corey_: during various smoke breaks |
| 21:02.09 | _Corey_ | ah :) |
| 21:02.20 | Qwell | I know they were in there for my talk though |
| 21:04.10 | _Corey_ | I didn't realize their project was that far along |
| 21:04.47 | Qwell | http://www.personal.psu.edu/wcs131/blogs/psuvoip/ |
| 21:05.03 | r0m|u | sounds like big project |
| 21:05.04 | _Corey_ | Univ. of Penn has about 8-10k phones on Asterisk/Kamailio as far as I know |
| 21:05.53 | *** join/#asterisk ndespres (~neil@208.103.80.226) |
| 21:05.58 | *** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au) |
| 21:06.58 | Qwell | _Corey_: upenn was >15000 as of 2010 |
| 21:07.16 | Qwell | according to a quick google search, anyways |
| 21:07.48 | _Corey_ | I think that number may have included some voicemail numbers |
| 21:07.54 | Qwell | maybe |
| 21:08.25 | _Corey_ | (They're one of our customers... we do several departments) |
| 21:08.36 | Qwell | ahh |
| 21:08.53 | _Corey_ | It's a long story, but they actually use some Switchvox now too |
| 21:09.20 | r0m|u | I which Rice would use Asterisk :( |
| 21:09.42 | r0m|u | whish* |
| 21:09.57 | r0m|u | wish* dam it |
| 21:10.00 | r0m|u | lol |
| 21:12.12 | brotherSand | Question: so is there a particular term for running multiple phone numbers down one voip line? |
| 21:13.18 | r0m|u | trunking? |
| 21:13.19 | _Corey_ | brotherSand: You'll get some divergent opinion on this one, but "trunking" is generally used to describe this (there are not really VoIP "lines" per se) |
| 21:15.31 | [TK]D-Fender | thinking there is a "term" is a bad start :) |
| 21:15.34 | brotherSand | Yeah, that's what makes it hard to search for I guess. |
| 21:15.36 | [TK]D-Fender | It just "is". |
| 21:15.46 | r0m|u | ~trunking |
| 21:15.50 | r0m|u | :/ |
| 21:15.52 | [TK]D-Fender | 1 account, multiple numbers in/out. |
| 21:16.02 | [TK]D-Fender | ~trunk |
| 21:16.02 | infobot | methinks trunk is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant |
| 21:16.16 | r0m|u | ah! |
| 21:16.22 | [TK]D-Fender | ^^^^ note the disqualification of "SIP" in a certain context |
| 21:17.53 | ndespres | Hi everyone, I'm new here, and new to Asterisk. One of my clients is having problems with inbound audio on their box, and I"m not even sure where to start looking |
| 21:18.06 | r0m|u | NAT/Firewall |
| 21:18.30 | r0m|u | those are usual symptoms of a NAT/Firewall in between |
| 21:18.53 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:18.55 | r0m|u | ndespres, ^^ |
| 21:19.07 | ndespres | The strange thing is that there is no config in their firewall at all that references the asterisk server.. makes me worried that someone forgot to 'wr mem'. |
| 21:20.05 | r0m|u | ndespres, looks like the ports are been blocked. forward the need it ports accordingly |
| 21:21.28 | r0m|u | 5060 for SIP and 1000 to what ever channels your need IE: 10000-10004 will give you two channels. 2 channels per each session |
| 21:22.07 | brotherSand | thanks all. I may be back. |
| 21:22.09 | r0m|u | better said it will give you two calls each calls uses 2 channels |
| 21:22.13 | *** part/#asterisk brotherSand (~da5id@c-98-228-227-159.hsd1.il.comcast.net) |
| 21:22.32 | ndespres | Thanks r0m|u. I'll start there. I forward those ports to the asterisk box, or to the IPs of the phones? (I don't even know yet what kind of phones they are) |
| 21:22.43 | ndespres | NOthing like diving in headfirst to a new system at 4pm on a Friday! |
| 21:23.30 | r0m|u | ndespres, Given that the phones talk to the asterisk system than The asterisk box. |
| 21:24.08 | r0m|u | use care and be security conscious. I only forward ports with a given source. I never open them to the world |
| 21:24.42 | r0m|u | also make sure that the asterisk box is secure properly with tools like fail2ban or iptables. |
| 21:25.37 | r0m|u | ndespres, was this system setup by somebody else? a consultant? |
| 21:26.17 | ndespres | Yes, my impression is that it's "some guy", friend of the company owner, who is conveniently out of town. |
| 21:26.38 | r0m|u | ouch. |
| 21:27.20 | *** join/#asterisk dailylinux (~test@88.87.48.115) |
| 21:29.05 | r0m|u | well you if you dont know how to swim the asterisk waters you are for a nice surprise :) though you shouldn't have to touch the system just the router/firewall |
| 21:30.44 | r0m|u | ndespres, good luck. there is good people here and some really good veterans that can definitely help. sucks though that you have to do this on a Friday. :/ |
| 21:30.51 | *** join/#asterisk BlackBishop (~dexter@ipv6.d3xt3r01.tk) |
| 21:31.02 | r0m|u | and by the way: |
| 21:31.05 | r0m|u | ~book |
| 21:31.06 | 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 |
| 21:31.57 | ndespres | Thanks. I ordered the book this week! Had another asterisk problem with another client that easy to solve, but I figured I should be prepared. |
| 21:35.46 | r0m|u | ndespres, good thinking. ;) |
| 21:41.47 | *** join/#asterisk brotherSand (~da5id@38.106.255.75) |
| 21:42.00 | r0m|u | brotherSand, that was fast lol |
| 21:42.13 | brotherSand | yeah, I'm fickle |
| 21:42.48 | r0m|u | :) |
| 21:43.17 | brotherSand | don't have a specific question at the moment, but I have a feeling I will in a bit. (reading "Definitive Guide" now) |
| 21:43.36 | r0m|u | cool |
| 21:44.20 | brotherSand | unless ... anyone here have any direct experience with Vitelity as a voip provider? |
| 21:46.08 | r0m|u | brotherSand, what is your question? |
| 21:47.01 | r0m|u | Vitelity is a good provider. |
| 21:47.12 | brotherSand | Well, I was just wondering if anyone has any warnings or cautions about using Vitelity. That appears to be the company we are going to use as our voip provider for our Buenos Aires office. |
| 21:47.28 | r0m|u | They are good as far as I know. |
| 21:47.43 | brotherSand | That's good to hear. |
| 21:47.47 | r0m|u | I been looking in to using them for some DID's in Puerto Rico. |
| 21:48.19 | _Corey_ | brotherSand: As long as you don't need phone support or anything mission-critical, they're fine |
| 21:48.30 | r0m|u | ^^ |
| 21:49.09 | brotherSand | Ah, this is why it's good to ask these things. _Corey_ , you're saying their support is a bit laid back? |
| 21:49.23 | brotherSand | (what does DID stand for ?) |
| 21:49.32 | r0m|u | ~did |
| 21:49.32 | infobot | from memory, did is Direct Inward Dialing, or just a phone number |
| 21:49.36 | _Corey_ | Doesn't exist as far as I know, aside from an e-mail helpdesk |
| 21:49.40 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 21:49.44 | r0m|u | _Corey_, They do |
| 21:49.46 | brotherSand | ah |
| 21:49.52 | r0m|u | echnical Support |
| 21:49.52 | r0m|u | support@vitelity.com |
| 21:49.52 | r0m|u | (888) 898-4835 |
| 21:49.57 | r0m|u | ^^ |
| 21:50.22 | _Corey_ | someone at Astricon was telling me they had a rough time trying to get help over the phone from them during an emergency |
| 21:50.28 | r0m|u | Though I did heard that there support is a bit on the laid back |
| 21:50.35 | r0m|u | side* |
| 21:50.48 | _Corey_ | they were told to open a ticket on the website... |
| 21:51.09 | r0m|u | _Corey_, I wouldnt doubt it. I heard similar situations but not emergency's |
| 21:51.10 | _Corey_ | I've used them for some termination for a while and a little origination with non-critical numbers |
| 21:51.39 | _Corey_ | They have capacity... I've thrown 80-100 simultaneous calls at them already |
| 21:51.46 | r0m|u | brotherSand, here are some of the more popular itsp |
| 21:51.58 | r0m|u | ~itsp |
| 21:51.58 | infobot | [~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs. |
| 21:52.35 | brotherSand | ah, that's a useful bot |
| 21:53.21 | brotherSand | needs an "~itsplist-sa" though for South America. :-) |
| 21:53.28 | r0m|u | _Corey_, I been looking for itsp that can serve PR but most consider PR long distance.... The only rate I found cheap enough was Vitality. Any ideas of one with better rates or PR as local? |
| 21:54.23 | brotherSand | Yes, our concern is a lot of Argentina <--> Chicago calls |
| 21:54.53 | r0m|u | brotherSand, you are in the same boat as I am. Looking for a competitive carrier for outside the us though PR is US most ITSP dont think so. |
| 21:55.16 | r0m|u | for now vitality is my only solution I think.... |
| 21:55.31 | Qwell | r0m|u: You'd be lucky to have hawaii and alaska be considered US |
| 21:55.40 | brotherSand | Hahahaha |
| 21:55.46 | r0m|u | LOL |
| 21:55.48 | Qwell | that wasn't a joke. |
| 21:55.53 | brotherSand | still funny |
| 21:55.57 | r0m|u | I know what you mean. |
| 21:55.57 | _Corey_ | r0m|u: Try Alcazar networks |
| 21:56.22 | _Corey_ | r0m|u: Best rates I've seen in a while |
| 21:56.34 | r0m|u | _Corey_, Thanks! |
| 21:56.42 | brotherSand | r0m|u: so you have settled on Vitality due to rates or is there some other motivating factor? |
| 21:57.17 | r0m|u | Just rates. at the end is all about money. well most of the time |
| 21:59.27 | jaytee | Flowroute Puerto Rico = $0.0113 |
| 21:59.32 | jaytee | per minute |
| 22:01.07 | r0m|u | Thanks jaytee. |
| 22:01.19 | r0m|u | I am out for the day. Going home. work is over |
| 22:01.26 | r0m|u | for today that is :) |
| 22:01.29 | r0m|u | cya guys! |
| 22:02.59 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-magbeooyjqcldqya) |
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| 22:04.55 | brotherSand | later r0m|u |
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| 22:51.40 | michael-i | Is it possible to disable individual modules via the command line when I run configure instead of using menuselect? |
| 22:55.58 | brotherSand | cheers |
| 22:55.59 | *** part/#asterisk brotherSand (~da5id@38.106.255.75) |
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