| 00:01.28 | citywok | yea idk, he didn't seem surprised nor apologetic and clarified their policy about suspending accounts whenever without warning, which isn't worth the potential headache for me. we're already working to close our bw.com account and move to qwest as our SIP carrier so closing both accounts at the same time isn't much extra work. |
| 00:02.04 | citywok | it's my fault for porting somewhat important phonenumbers to a tier3 carrier instead of going with a tier1 carrier in the first place |
| 00:02.31 | citywok | i actually don't really care about the outbound being blocked, it's the shutting off all my inbound numbers that is the problem. |
| 00:08.16 | *** join/#asterisk MattJ100 (~MattJ@82.153.139.216) |
| 00:09.04 | Pio | are there any lists of analog telephony interfaces that are asterisk friendly? like so I can get my POTS line hooked up to an asterisk box.. this little guys the closest ive seen http://www.sangoma.com/products/hardware_products/analog_telephony/usb_fxo.html |
| 00:09.24 | Pio | usb preferred.. no pci card slots left in the box i wanna use |
| 00:09.48 | [TK]D-Fender | Pio: How many lines? what kind of call voluem? What kind of use (home/business), etc? |
| 00:09.57 | Pio | one line, home use, low volume |
| 00:10.12 | *** join/#asterisk deltalytic (~haryv@154.5.144.132) |
| 00:11.04 | [TK]D-Fender | Pio: that device could be fine, but I'd recommend the Linksys SPA-3102 |
| 00:11.08 | Pio | i just am trying to meld together my local land line and SIP.. so people can still call my local number and ring my phones, but people could also call my SIP numbers and i could make outgoing SIP calls on the same phones |
| 00:11.24 | [TK]D-Fender | Pio: its an Etherenet device that will speak to * over your LAN and also supports 1 phone. |
| 00:11.34 | deltalytic | I have a ip500 that is looping in bootmode. Getting a error of config file error error is 0x0. I have not made any changes to sip.ld. I did make a minor change to one of the mac.cfg files such as a server address change. Any ideas? |
| 00:12.01 | deltalytic | file name is also lowercasew |
| 00:12.01 | deltalytic | lowercase |
| 00:12.25 | deltalytic | Or, can download ready to go ip500 config files and make some minor changes :) |
| 00:12.45 | MattJ100 | Can anyone give me a pointer on how to make mailbox selection automatic (dependent on the calling extension) to VoicemailMain? |
| 00:12.53 | [TK]D-Fender | Pio: Also in the case of a power failure, it will bridge the phone & line ports together so you can failover. |
| 00:12.54 | [TK]D-Fender | deltalytic: they can loop if youscrew up the formatting on your configs |
| 00:13.08 | deltalytic | possibly |
| 00:13.25 | MattJ100 | I tried ${EXTEN}, but I think that gives the called extension, not the calling |
| 00:13.26 | Pio | [TK]D-Fender, wow that's awesome |
| 00:13.39 | deltalytic | I will remember to put a notepad next to me and write down the changes. |
| 00:13.55 | [TK]D-Fender | MattJ100: and what holds the NUMBER of the CALLER so that you can ID them? |
| 00:14.22 | MattJ100 | [TK]D-Fender: That's what I'd like to find out :) They're a local user, so I'm guessing it's somewhere... no? |
| 00:14.58 | [TK]D-Fender | MattJ100: read what I said a few dozen more times... |
| 00:14.59 | deltalytic | by default, is 1.6 almost verbose silent? Get no errors popping up in cli..ever. |
| 00:15.15 | [TK]D-Fender | deltalytic: depends. what is your verbose set to? |
| 00:15.22 | deltalytic | not sure. |
| 00:15.37 | deltalytic | what do I type or is it pre configured? |
| 00:15.38 | [TK]D-Fender | delGo look |
| 00:15.46 | [TK]D-Fender | deltalytic: go CHANGE the verbose level |
| 00:15.55 | [TK]D-Fender | deltalytic: and it will tell you what it was at before |
| 00:16.37 | MattJ100 | [TK]D-Fender: Didn't help |
| 00:16.50 | deltalytic | got it set to 3 now |
| 00:17.14 | [TK]D-Fender | MattJ100: CALLERID <----------------- |
| 00:17.47 | deltalytic | also need to get 5060/10000 though this blasted telus router |
| 00:29.40 | deltalytic | shees! 638 pages for a home router manual? |
| 00:29.59 | deltalytic | I can see for gone with the wind but for a router manual? |
| 00:31.00 | deltalytic | my router is mangaling the packets. |
| 00:31.04 | deltalytic | <PROTECTED> |
| 00:31.04 | deltalytic | Really destroying SIP dialog '3529bd87495448fe57e4ebf61929d003@192.168.1.2' Method: REGISTER |
| 00:31.07 | deltalytic | hehe |
| 00:35.46 | [TK]D-Fender | deltalytic: Doesn't look mangled.... |
| 00:37.36 | deltalytic | some of the users say so in some of the forums. Anyway, see what else I can find on this unit. |
| 00:41.59 | [TK]D-Fender | deltalytic: Most members of most forums are total idiots... |
| 00:46.13 | deltalytic | yea I can see |
| 00:46.19 | deltalytic | thay do not respond |
| 00:46.40 | deltalytic | BTW, I put my asterisk box on the dmz and stil, cannot register |
| 00:46.54 | deltalytic | Seems other people are having sip passthough issues on this router. |
| 00:49.17 | deltalytic | TK, ever hear of anyone putting a firewall on a asterisk box with no security issues? |
| 00:49.39 | deltalytic | might just do that and put the box between the dsl and this d-link telus router. |
| 00:51.38 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 00:52.09 | WIMPy | Issues like being able to switch off the firewall, once the asterisk backdoor has been found? |
| 00:52.27 | deltalytic | :) |
| 00:52.36 | [TK]D-Fender | deltalytic: I did that myself for years. |
| 00:52.55 | deltalytic | Just cuts down on space and power |
| 00:52.56 | [TK]D-Fender | deltalytic: Tecnically less secure for being that much closer, but its a dumb as you are :) |
| 00:53.01 | *** join/#asterisk m_tadeu (~quassel@89.180.196.93) |
| 00:53.22 | deltalytic | I know its pushing the limits of risk |
| 00:53.29 | [TK]D-Fender | deltalytic: I had a Sangoma S518 ADSL PCI card in my server direct to plug to the wall |
| 00:53.42 | [TK]D-Fender | delnothing external, and dual GB NIC's inside |
| 00:54.19 | deltalytic | I do have some other non vendor specific routers laying around. |
| 00:54.22 | WIMPy | is alo using only one PC for everything. |
| 00:54.32 | WIMPy | s/alo/also/ |
| 00:56.07 | *** join/#asterisk rrb3942 (~rbullock@cpe-67-242-215-62.rochester.res.rr.com) |
| 00:57.21 | [TK]D-Fender | deltalytic: its not really dumb. Dumb would be using a Windows box for this :) |
| 00:57.39 | [TK]D-Fender | deltalytic: Yes its more exposed but if you're smart with IPTABLES you'll be fine |
| 00:58.34 | deltalytic | hhe |
| 00:58.38 | deltalytic | like my parents. |
| 00:58.43 | deltalytic | thay do not understand |
| 00:58.52 | deltalytic | but also, there is no data on there system |
| 00:59.23 | deltalytic | time to learn iptables ;) |
| 01:01.43 | deltalytic | telus issues. "please hang up and dial again" crap phone service |
| 01:05.08 | deltalytic | ugg, wife wanted to switch from shaw to telus. Its worse. BTW, I have a OLD digium card that needs to be chucked and get one that has no echo and no volume issues. |
| 01:05.54 | deltalytic | brb, rebooting router |
| 01:10.02 | MattJ100 | [TK]D-Fender: Thanks - I guess that's my issue... ${CALLERID(num)} doesn't seem to yield anything |
| 01:12.18 | *** join/#asterisk esaym (~esaym153@cpe-70-120-95-19.satx.res.rr.com) |
| 01:12.32 | esaym | howdy, is there any faq's for migrating from 1.4 to 1.8? |
| 01:13.33 | WIMPy | esaym: Yes in the UPGRADE*.txt. |
| 01:13.43 | esaym | hmm, is that all? ok |
| 01:13.44 | esaym | ty |
| 01:16.33 | WIMPy | esaym: Well, there's no questions, but it's supposed to tell you what you need to be aware of. |
| 01:17.21 | *** part/#asterisk Get_The_Fish (~Get_The_F@c-24-8-50-199.hsd1.co.comcast.net) |
| 01:21.44 | esaym | hmm, looks like I can for from 1.4 to 1.8 without changing anything |
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| 01:41.20 | Bloudermilk | Any AGI pros out there? |
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| 02:08.19 | Weezey | I've got 1.8 running but I'm getting a bunch of errors on the console |
| 02:09.06 | Weezey | looks like an IPv6 name resolution problem: ERROR[18979]: netsock2.c:245 ast_sockaddr_resolve: getaddrinfo("", "(null)", ...): Name or service not known |
| 02:09.25 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 02:09.39 | Weezey | and for some reason none of my iax realtime peers are showing up |
| 02:10.05 | Weezey | NOTICE[18974]: chan_iax2.c:7955 register_verify: No registration for peer 'username' (from IP address) |
| 02:10.18 | Weezey | sip peers look like they work just fine though. |
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| 03:25.39 | nsgn | what command may there be to quickly view the codec currehtly in use for my main trunk? |
| 03:41.51 | Weezey | sip show channels |
| 03:42.54 | Weezey | iax refuses to work from realtime static and realtime iaxpeers/iaxusers so I'm going to a flat file until I can figure it out. but now I'm getting: reg_source_db: IAX/Registry astdb host:port invalid - xxx.xxx.xxx.xxx:4569 |
| 03:44.20 | nsgn | Weezey, thx :) |
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| 04:27.00 | *** part/#asterisk Defraz (~Defraz@c72co-edge-router.fuzecore.com) |
| 04:29.15 | *** join/#asterisk dzup1 (~alex@unaffiliated/dzup) |
| 04:42.39 | *** join/#asterisk Montys (~dmartinez@nat/digium/x-pxijmeqnkcscdrqr) |
| 04:42.43 | Montys | elo |
| 04:43.09 | Montys | Do you know if there is any way that I can change a SIP header (FROM) in asterisk |
| 04:46.35 | [TK]D-Fender | Montys: to what? |
| 04:47.18 | *** join/#asterisk BeeBuu (~chatzilla@58.252.73.164) |
| 04:47.44 | BeeBuu | is there 3rd edition book about the asterisk? |
| 04:48.00 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-fuofxzhbgzbnlntf) |
| 04:48.27 | [TK]D-Fender | BeeBuu: It is going to be release very soon |
| 04:48.52 | BeeBuu | Oh,that's good news |
| 04:49.15 | BeeBuu | about asterisk 1.8? |
| 04:49.42 | [TK]D-Fender | yes |
| 04:49.43 | Montys | [TK]D-Fender: FROM THIS: From: "anonymous" <sip:anonymous.invalid>;tag=1234567890-1234567890 TO: From: "Asterisk" <sip:MyAsterisk>;tag=1234567890-1234567890 |
| 04:49.58 | [TK]D-Fender | Montys: What call? |
| 04:50.40 | Montys | [TK]D-Fender: I'm not sure what you asking |
| 04:51.03 | [TK]D-Fender | Montys: where is the call from/to? |
| 04:51.14 | Montys | FROM |
| 04:51.27 | [TK]D-Fender | Montys: then no. |
| 04:51.32 | BeeBuu | ~book |
| 04:51.33 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 04:52.10 | Montys | [TK]D-Fender: I want to remove the \"anonymous" <sip:anonymous.invalid>\ and put other thing |
| 04:52.19 | Montys | [TK]D-Fender: Thanks, |
| 04:53.13 | Montys | [TK]D-Fender: So, currently asterisk does not have ability to modify the FROM header of a SIP call.... Thanks for your help :) |
| 04:54.49 | [TK]D-Fender | Montys: there is no point in "removing" a header. * is not a PROXY. It doesn't FORWARD anything |
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| 06:32.24 | awk | Hi, Is it possible up the the TX and RX gain on a SIP trunk. I have a VoIP connection to a supplier and it's very soft. When I used a Quintum I was able to use a TX and RX gain on the IP Routing Group of that gateway ? |
| 06:33.26 | kaldemar | awk: core show function VOLUME |
| 06:35.44 | awk | 1.4 Asterisk |
| 06:38.10 | kaldemar | awk: VOLUME was introduced in 1.6.0. |
| 06:39.44 | awk | blaaaah |
| 06:55.44 | ChannelZ | Welcome to 2010 |
| 07:00.49 | shamelessn00b | hey can anyone help me out in discovering the jack applicaltion |
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| 07:07.35 | ChannelZ | core show application drop pants |
| 07:10.19 | shamelessn00b | :p |
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| 07:20.42 | ChannelZ | ^^^^ |
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| 07:27.05 | awk | I guess it's getting time for me to cut over.. Just such a head ache with stability. It took 1.2 how many releases? 1.4.36 is buggy and doesn't hang up channels correct.. If a call comes into a queue -> agent XYZ tranfers to Exten -> ABC, agent XYZ can't get a call till Exten ABC hangs up. So I am so afraid to start using 1.6 and land up for the next 2 years with unhappy clients |
| 07:27.36 | awk | How do you guys find the stability of 1.6, using many of the features in a very busy env. |
| 07:27.47 | awk | EG: 20k calls inbound to a call centre per day |
| 07:28.31 | Montys | well, As far as I understand 1.6.2 seems stable |
| 07:28.47 | Montys | awk; you dont want to use asterisk 1.6.0 or 1.6.1 |
| 07:30.01 | shamelessn00b | ChannelZ: I wanted to set Jack as an endpoint for the call (block dialplan execution) and then stream the channel audio to a java server |
| 07:30.07 | kaldemar | 1.2, 1.4 and 1.6.X are not completely comparable in that sense because of the development process changes, in my opinion. |
| 07:30.27 | *** part/#asterisk c0rnoTa (~c0rnoTa@178.177.66.41) |
| 07:31.11 | shamelessn00b | right now I'm using EAGI for that purpose |
| 07:31.28 | shamelessn00b | thing is, I dont have seperate audio streams in EAGI |
| 07:31.44 | shamelessn00b | for audio coming into the channel and going out of the channel |
| 07:34.44 | Montys | kaldemar: yeah -- You cannot compare one branch to another. I honestly thing that the only why that you could check for stability would be to check it by yourself |
| 07:37.42 | *** join/#asterisk schmidts (~schmidts@lmlo.sil.at) |
| 07:37.46 | schmidts | good morning |
| 07:38.34 | BeeBuu | hi,schmidts |
| 07:38.48 | ChannelZ | What do chineese people call a "lazy suzan"? |
| 07:40.00 | ChannelZ | shamelessn00b: hmm no idea, never even heard of the jack app |
| 07:41.08 | shamelessn00b | the only difference is that it blocks dialplan execution whereas jack_hook is non blocking |
| 07:41.23 | shamelessn00b | http://www.voip-info.org/wiki/view/Asterisk+cmd+jack |
| 07:43.11 | awk | Thanks guys... Let me give 1.6.2 a test at the office and then move out from there.. |
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| 09:53.38 | *** join/#asterisk nik_ (c636cac3@gateway/web/freenode/ip.198.54.202.195) |
| 09:54.48 | nik_ | hi everyone |
| 09:54.55 | nik_ | please help me with "Channel 0/1, span 1 got hangup, cause 81" |
| 09:55.30 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 09:55.34 | nik_ | apparently cause 81 is defined as AST_CAUSE_INVALID_CALL_REFERENCE |
| 09:56.25 | nik_ | I've got this happening randomly, some calls get through just fine, other get this error |
| 09:56.30 | nik_ | it happens very often |
| 09:56.41 | nik_ | maybe 30-40% of the time |
| 10:02.27 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 10:07.23 | WIMPy | nik_: If it's a rare thing it could just be a race condition. |
| 10:07.38 | WIMPy | Like answering a call that was dropped in the same moment. |
| 10:09.35 | nik_ | hi again WIMPy |
| 10:09.38 | nik_ | no |
| 10:09.47 | nik_ | it is very frequent condition |
| 10:09.57 | nik_ | I just replicated it few moments ago |
| 10:10.23 | WIMPy | I can't remember what it was exactely, but I know I've seen signalling that was obviousely senseless, because of timing issues. |
| 10:10.43 | WIMPy | Ok, when exactely does it happen? |
| 10:12.04 | nik_ | random |
| 10:12.05 | nik_ | but very often |
| 10:12.34 | nik_ | like I said, 30-40% of calls at the moment |
| 10:12.42 | WIMPy | Incoming? outgoing? When answering? At other r |
| 10:12.47 | WIMPy | oops |
| 10:12.50 | WIMPy | Incoming? outgoing? When answering? At other times? |
| 10:13.37 | *** join/#asterisk vallhalla (~dave@212.183.140.39) |
| 10:13.38 | nik_ | when answering |
| 10:13.42 | vallhalla | hello all |
| 10:13.49 | nik_ | only when answering |
| 10:14.17 | nik_ | when you pick up the phone, you hear "idle" but the display shows "talking", while the other party hears "ringing" tone |
| 10:15.54 | WIMPy | tries to remember. |
| 10:16.09 | WIMPy | That mighte be the one I was thinking of. |
| 10:16.50 | WIMPy | I think it was something about channel allocations. But that (alone) would not explan the invalid callref. |
| 10:17.07 | WIMPy | Do you have a intense trace of such an event? |
| 10:18.55 | nik_ | no, let me try enable this and see the next call |
| 10:20.25 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 10:21.01 | WIMPy | Actually to full console output may be helpfull, including the pri debug and the cores call handling hessages. |
| 10:22.55 | TimeRider | .. I find tail -f /var/log/asterisk/full useful |
| 10:23.37 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 10:24.35 | WIMPy | Yes, if you have it anabled :-) |
| 10:25.16 | nik_ | http://pastebin.com/teJ2SMpU |
| 10:25.32 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 10:25.32 | TimeRider | WIMPy: is default not full logging to there? |
| 10:25.50 | nik_ | WIMPy: this is just some stuff that keep happening, not sure if it is related to the problem, but there is some timer problems it seems |
| 10:26.11 | *** join/#asterisk xheliox (jeff@pdpc/supporter/student/xheliox) |
| 10:26.29 | WIMPy | TimeRider: No and it can actually be harmfull. I've seen an installation that stalled, because the harddisc couldn't handle the amount of data. |
| 10:26.50 | TimeRider | Yeah, it does create MASSIVE logs like |
| 10:27.39 | TimeRider | so... me now assuming that I'm not seeing FULL logs in full, where do we set the level? I just leave mine default and see a LOT in there |
| 10:28.07 | WIMPy | nik_: No, that looks perfectely normal. |
| 10:28.38 | WIMPy | TimeRider: logger.conf |
| 10:29.01 | nik_ | WIMPy: ok |
| 10:29.19 | nik_ | WIMPy: I will try get something more |
| 10:29.33 | TimeRider | THX :) |
| 10:55.43 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 10:56.27 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:03.59 | *** join/#asterisk jeremy_g (~jeremey_g@gw-sthlm01.rebtel.com) |
| 11:04.41 | *** join/#asterisk MattJ (~MattJ@82.153.139.216) |
| 11:05.29 | *** join/#asterisk BenC[UK] (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
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| 11:06.50 | nik_ | WIMPy: http://pastebin.com/Rhw2Acg6 |
| 11:07.11 | nik_ | please have a look at this, it might contain some unrelated stuff, but I think I managed to cut out the failed call portion more or less |
| 11:13.09 | *** join/#asterisk BenC[UK]_ (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 11:26.16 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 11:28.09 | WIMPy | nik_: The seem to be something missing. |
| 11:28.38 | WIMPy | But It looks like you have some TEI management issues with power saving. |
| 11:29.30 | WIMPy | Hase there been RRs going back and forth prior to the call? |
| 11:30.54 | WIMPy | Are you connected to a public networ ot to another PBX? Just wondering about a global call. |
| 11:35.13 | *** join/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:35.44 | *** part/#asterisk rossand (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 11:42.20 | *** join/#asterisk oDesk (~f@94.97.73.73) |
| 11:42.28 | oDesk | hello, |
| 11:42.57 | oDesk | can i limit one of the dialplans/context to 5 minutes then hangup ? |
| 11:47.58 | nik_ | WIMPy: what is RRs? |
| 11:52.43 | nik_ | WIMPy: the same network is used for internet and general LAN, no other PBX |
| 11:54.29 | eMBee | good evening |
| 11:54.52 | eMBee | what is WARNING chan_dahdi.c: Channel '1' ignored. trying to tell me? |
| 11:55.10 | eMBee | i get this for all 3 challes that i have |
| 11:56.23 | *** join/#asterisk CoderForLife (~Miranda@cpe-174-103-152-61.cinci.res.rr.com) |
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| 12:01.53 | *** join/#asterisk schmrz (~schmrz@usaidgbep1-178.bih.net.ba) |
| 12:02.10 | schmrz | Hello Everyone! |
| 12:03.16 | schmrz | I'm currently googling, but can anyone recommend a c++ api for interfacing with Asterisk ? |
| 12:05.30 | BenC[UK] | schmrz: interfacing how? |
| 12:05.35 | BenC[UK] | administration? |
| 12:05.36 | BenC[UK] | or calls? |
| 12:07.14 | schmrz | Calls. I need to receive Asterisk events and also to be able to send actions. Something like Asterisk.NET for C#. |
| 12:10.34 | *** join/#asterisk netrat (~netrat@va-69-34-33-233.sta.embarqhsd.net) |
| 12:11.37 | schmrz | BenC[UK]: Any suggestions ? |
| 12:13.38 | drmessano | You can use AMI to interface with Asterisk |
| 12:14.43 | schmrz | drmessano: I need to do it programatically from within my application. |
| 12:14.56 | russellb | http://matt-land.com/astxx/ |
| 12:15.28 | drmessano | Um, schmrz, that's why AMI exists |
| 12:15.40 | schmrz | russellb: Thank you, I saw that one. |
| 12:15.56 | tzafrir | eMBee, is this on configuration reload? |
| 12:17.09 | drmessano | schmrz, AMI is well documented and you should be able to do anything you need in a few lines of code.. I mean, people manipulate AMI with shell scripts |
| 12:17.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@216.191.106.163) |
| 12:19.41 | WIMPy | nik_: Receive Ready. |
| 12:20.05 | schmrz | Also, can anyone tell me if it's a good idea to build something like a softphone but using only Asterisk ? I have built an application that handles incomming, outgoing calls, call transfers and everything that a normal softphone does. But it felt like I'm raping Asterisk in the process of writting my application... It's kinda late now, because the application is already in production but I was wondering if someone had similar experiences and co |
| 12:20.07 | schmrz | uld share a few tips with me ? |
| 12:20.14 | shamelessn00b | what dialplan app does app_talkdetect.c. provide? |
| 12:20.18 | schmrz | drmessano: My application is actually a softphone, but using Asterisk only. |
| 12:20.29 | schmrz | drmessano: I can't really do what I want to with shell scripts. |
| 12:20.30 | *** join/#asterisk puzzled (~foobar@puzzled.xs4all.nl) |
| 12:20.48 | puzzled | hi |
| 12:21.48 | [TK]D-Fender | schmrz: * is a PBX toolkit, not a softphone toolkit. * SUCKS as a "phone |
| 12:22.12 | [TK]D-Fender | schmrz: Poor choice as a base. Thre are tons of PROPER and easy libs out there to build clients with |
| 12:22.21 | drmessano | schmrz, I wasn't suggesting shell scripts. I was pointing out that you don't need another layer on top of AMI to use AMI within your favorite language. AMI is SIMPLE to use. |
| 12:22.34 | russellb | disagrees with drmessano |
| 12:22.42 | russellb | a library to make the AMI feel more native to the language is handy. |
| 12:22.47 | russellb | not needed, but nice to have. |
| 12:22.51 | [TK]D-Fender | schmrz: So unless you're doing this just to prove you can, then you are picking the wrong product for the job |
| 12:23.16 | drmessano | russellb, if it exists, and is up to date.. if not, it's far from a dealbreaker, and nothing you can't do yourself |
| 12:23.18 | russellb | also disagrees with [TK]D-Fender and things it would be easy to build a phone on asterisk, and that it's a powerful base |
| 12:23.26 | russellb | drmessano: indeed |
| 12:23.37 | nik_ | WIMPy: how do I check the RR? |
| 12:24.12 | drmessano | 10 ASTERISKCALL "8675309" |
| 12:24.16 | drmessano | 20 GOTO 10 |
| 12:24.17 | drmessano | RUN |
| 12:24.23 | drmessano | Asterisk BASIC wardialer |
| 12:24.24 | russellb | depends what you want to make ... another boring basic softphone? sure, go use a simple SIP library ... |
| 12:24.34 | schmrz | [TK]D-Fender: That's what I thought... The problem is that I can't use the SIP protocol. Because the client demanded that he uses the application in combination with his phone. |
| 12:24.35 | [TK]D-Fender | russellb: And you could use a Hummer as a golf cart... doesn't make it a good idea. |
| 12:24.44 | russellb | if you want to make something awesome, with features nobody else has put into a softphone, i like asterisk is an awesome choice to make it easy to build powerful features |
| 12:24.51 | russellb | where's your vision! |
| 12:25.04 | shamelessn00b | app_talkdetect.c what module does it provide me?????? |
| 12:25.13 | schmrz | russellb: Yeah, but handling calls is not easy. |
| 12:25.21 | schmrz | russellb: And everything else |
| 12:25.45 | russellb | well, AMI is what you've got for that. |
| 12:25.50 | schmrz | for example, when you make a blind transfer from the phone you can't really know that the user did that |
| 12:25.54 | russellb | anyway, *goes back to writing this book* |
| 12:26.13 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:26.13 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:26.30 | drmessano | schmrz, I am pretty sure russellb is familiar with asterisk programming |
| 12:26.36 | drmessano | I would hope so anyway |
| 12:26.43 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:26.55 | russellb | drmessano: i submitted a patch once |
| 12:27.08 | schmrz | I'm not saying that he is not. But I'm not programming asterisk, I'm programming *with* asterisk. |
| 12:27.18 | russellb | understood |
| 12:27.35 | russellb | AMI was not designed for that ... but it can be (ab)used for it |
| 12:27.52 | russellb | there should be events that let you know a transfer has occurred ... |
| 12:27.59 | schmrz | Exactly. That's why I said that it felt like I'm *raping* asterisk while writting my application :) |
| 12:28.10 | russellb | heh |
| 12:28.14 | WIMPy | nik_: You see them contantly going back and forth. Like what you sent in your first pastebin. |
| 12:28.32 | russellb | if you're a C++ programmer, you could write a new control interface module, heh |
| 12:28.36 | zamba | is it possible to get a gsm interface on asterisk? |
| 12:28.40 | russellb | it really depends on your requirements ... |
| 12:28.43 | zamba | for routing calls to mobile phones through that? |
| 12:28.45 | russellb | zamba: search for chan_datacard |
| 12:28.50 | russellb | and chan_mobile |
| 12:29.05 | drmessano | I was going to try Asterisk 1.8 yesterday, but I can't find Asterisk.exe in the targa gzip thingo. |
| 12:29.13 | drmessano | Guess I need to give Linux another shot |
| 12:29.20 | *** join/#asterisk mintos (~mvaliyav@114.143.164.151) |
| 12:29.20 | zamba | russellb: ah.. thanks |
| 12:29.24 | WIMPy | zamba: As SIP gateways, as PCI cards or for USB sticks. |
| 12:29.47 | drmessano | chan_mobile is the best way to interface with a phone, IMO |
| 12:30.07 | zamba | WIMPy: hm? |
| 12:30.12 | leifmadsen | drmessano: I think you can use cygwin for that |
| 12:30.22 | WIMPy | correction: zamba: As SIP gateways, as PCI cards, for USB sticks or via a phone. |
| 12:30.25 | russellb | leifmadsen: i wonder if that still works ... probably not |
| 12:30.34 | leifmadsen | russellb: oh I'm certainly sure it won't ;) |
| 12:30.45 | zamba | WIMPy: i don't get your context.. what are you talking about? was that a question or a statement? |
| 12:30.49 | leifmadsen | russellb: if it works, I'd give you a case of beer |
| 12:30.54 | *** part/#asterisk netrat (~netrat@va-69-34-33-233.sta.embarqhsd.net) |
| 12:30.59 | russellb | it's not worth my time to even try |
| 12:31.08 | leifmadsen | russellb: if it doesn't work, then I don't have to pay for my kindles |
| 12:31.10 | russellb | i'd rather buy you a case of beer than try |
| 12:31.11 | drmessano | leifmadsen, I should be able to run Asterisk under Windows Services For UNIX. I am disappointed it's not supported and tested |
| 12:31.12 | WIMPy | zamba: It was the answer to your question. |
| 12:31.15 | russellb | leifmadsen: bitch |
| 12:31.25 | leifmadsen | russellb: that is also acceptable -- I'll expect a case of beer at astricon |
| 12:31.26 | WIMPy | zamba: Lots of options. |
| 12:31.32 | zamba | WIMPy: ok |
| 12:31.45 | russellb | leifmadsen: i can't write worth a crap today |
| 12:31.47 | leifmadsen | drmessano: maybe the big announcement at astricon is that asterisk now runs under WinNT ? |
| 12:31.48 | zamba | WIMPy: sip gateways are providers that deliver gsm traffic over sip, right? |
| 12:32.11 | leifmadsen | russellb: welcome to the club... I think there is something wrong with the schema for LDAP because I can't seem to show the last two values in the list of attributes |
| 12:32.23 | leifmadsen | I've asked on #openldap but I don't expect much of a response |
| 12:33.57 | drmessano | leifmadsen, it wasn't the 1.8 release you guys lied to me about through your teeth, because someone did that before Astricon. So whatever it is, is bigger than 1.8. |
| 12:34.24 | russellb | drmessano: dun dun dunnnn |
| 12:34.28 | leifmadsen | drmessano: or maybe we just got antsy and Kevin's speech will be anti-climatic? :) |
| 12:34.45 | drmessano | Ok, "lied" is a strong word, but it was strongly suggested announcing 1.8 at Astricon would be 'the big thing'. I knew that wasn't it. |
| 12:34.50 | drmessano | LOL |
| 12:34.56 | leifmadsen | maybe it still is! |
| 12:35.01 | russellb | leifmadsen: you're totally fired for releasing early |
| 12:35.03 | leifmadsen | and who suggested that? :) |
| 12:35.08 | leifmadsen | russellb: you told me to! |
| 12:35.12 | leifmadsen | blames marketing |
| 12:35.12 | russellb | SHH |
| 12:35.22 | drmessano | Wait |
| 12:35.26 | leifmadsen | they totally set the time on the bomb |
| 12:35.30 | drmessano | The acquisition rumours |
| 12:35.32 | drmessano | That must be it |
| 12:35.33 | leifmadsen | we had to release because of the announcment |
| 12:35.44 | leifmadsen | drmessano: LME is getting acquired and I will become rich! |
| 12:35.44 | drmessano | Digium DID get acquired! |
| 12:35.59 | leifmadsen | LME acquired Digium? I don't have those kinds of funds |
| 12:36.01 | russellb | drmessano: by the government of China |
| 12:36.03 | tzanger | hmm I should think about moving up from 1.4.23.1 |
| 12:36.08 | leifmadsen | tzanger: lol |
| 12:36.16 | leifmadsen | tzanger: have you used OpenLDAP at all? |
| 12:36.40 | tzanger | years ago, yes. I even bought a book about LDAP. I got bogged down in how to design the schema though and things didn't progress from there |
| 12:36.55 | drmessano | russellb, if you guys let Cisco or some other douche outfit acquire our beloved Digium, I am switching to Microsoft Lynx Server immediately |
| 12:37.25 | leifmadsen | tzanger: ah, the schema is the same part I am having issues with :) |
| 12:37.31 | leifmadsen | tzanger: trying to debug what is wrong |
| 12:37.34 | russellb | drmessano: we're going to let kevin announce 1.8.0.1, codename "oops" |
| 12:37.41 | leifmadsen | lol |
| 12:37.47 | leifmadsen | :|\ |
| 12:38.08 | tzanger | russellb: haha |
| 12:38.23 | WIMPy | zamba: No, I was referring to hardware. |
| 12:38.53 | WIMPy | zamba: Something like a channelbank. |
| 12:39.10 | zamba | WIMPy: ok.. but what is the cheapest option? |
| 12:39.18 | zamba | WIMPy: but still reliable? |
| 12:40.15 | WIMPy | The cheapest is probably an USB ttick or a left over phone. |
| 12:40.17 | [TK]D-Fender | [08:31]<zamba>WIMPy: sip gateways are providers that deliver gsm traffic over sip, right? <- NO. A SIP gateway is a BOX that * will speak to using SIP and have GSM ANTENNAS inside of them |
| 12:40.34 | [TK]D-Fender | zamba: And translate the SIP call into a GSM call |
| 12:40.55 | WIMPy | Hopefully not inside. |
| 12:41.40 | zamba | WIMPy: ttick? |
| 12:41.51 | zamba | stick? |
| 12:41.51 | zamba | hehe |
| 12:42.43 | WIMPy | zamba: The ones you use for UMTS connections. Some of them can do voice. |
| 12:42.58 | *** join/#asterisk bip (~bip@unaffiliated/bip) |
| 12:43.29 | bip | hello can I ask question about the asterisk GUI here ore there is a specific channel for that ? |
| 12:43.38 | bip | s/ore/or |
| 12:43.42 | zamba | WIMPy: http://www.voip-info.org/wiki/view/chan_mobile they are the once listed under usb bluetooth dongle? |
| 12:44.22 | WIMPy | Bluetooth and GSM (or UMTS) are NOT the same. |
| 12:44.37 | *** join/#asterisk GhOnDiE (~GhOnDiE@92.29.187.254) |
| 12:45.01 | *** part/#asterisk schmrz (~schmrz@usaidgbep1-178.bih.net.ba) |
| 12:45.14 | bip | http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect111_tt1345.html#id4148514 |
| 12:46.13 | nik_ | WIMPy: but what should I do about it? |
| 12:46.20 | bip | that link above says that every change done in the .conf file will show in the GUI, now i using some hand made .conf file but they do not show in the GUI, any hint will help ... |
| 12:46.38 | russellb | bip: depends which file ... the GUI primarily uses users.conf |
| 12:47.03 | dandre | hello, |
| 12:47.12 | dandre | I use asterisk 1.6.2. |
| 12:47.13 | bip | well i just edite sip.conf and extension.conf russellb |
| 12:47.42 | russellb | k, the only page in the GUI that will show that is the config file editor page ... the GUI doesn't mess with those for the most part |
| 12:47.53 | dandre | If I do a manager command 'core restart when convenient', asterisk stops bu never restart |
| 12:48.03 | WIMPy | nik_: Tell us more or paste a complete log of a missed call, including some time before it happens. |
| 12:48.08 | dandre | Is there any issue like this? |
| 12:48.31 | zamba | WIMPy: so basically just an old nokia phone, usb connectivitity to the server and that application/driver? |
| 12:48.46 | bip | you mean there is non way to create extensions and dialplans from the GUI russellb ? or to see those createdmanually in the GUI ? |
| 12:49.27 | WIMPy | zamba: Check the docs for chan_mobile. I haven't tried that. |
| 12:49.43 | russellb | i'm not a GUI expert. you create users and you create trunks ... you don't write custom dialplan |
| 12:49.53 | russellb | if you want to write custom dialplan, then no GUI provides much value |
| 12:50.00 | russellb | the whole point is to abstract it away into simpler concepts |
| 12:51.45 | bip | I need to write a dialplan nothing fancy, and I am a asterisk newbie, I can handle dealin with text files, but I have bem requested to fgure an easier way so people here can do routine tasks while I m not around |
| 12:52.15 | leifmadsen | bip: what kind of routine tasks? |
| 12:52.30 | nik_ | WIMPy: updated http://pastebin.com/qbjcJBBt, call starting at line +/- 1890 |
| 12:52.35 | bip | lÃike adding users, changing ring groups or messages |
| 12:52.50 | bip | tweakin the dialplan slightly |
| 12:53.14 | leifmadsen | bip: well tweaking the dialplan is not a routine task. That sounds like a documentation problems. |
| 12:53.39 | leifmadsen | I'm not sure how else you could do that, other than to write some sort of GUI :) |
| 12:53.45 | leifmadsen | (which is significantly more work) |
| 12:54.02 | bip | well do you know if there is any GUI specific documentation addressed to the end user ? |
| 12:54.13 | leifmadsen | I don't understand the question |
| 12:54.22 | bip | sorry |
| 12:54.32 | bip | lemme rephrase |
| 12:55.37 | bip | the asterisk book has a whoel chapter about GUI but sounds like it s adressing developers of customzed GUI, I was wondering if there was some sort of asterisk GUI end user manual available |
| 12:56.57 | russellb | not that I know of. |
| 12:57.11 | russellb | the next version of that book is going to have even less information about the GUI. :-p |
| 12:57.20 | bip | wonderful |
| 12:57.28 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 12:57.32 | bip | oh well is understendable |
| 12:58.19 | leifmadsen | especially since the focus of the book is asterisk administration as the lowest level |
| 12:58.22 | *** join/#asterisk BenC[UK] (~bencummin@cpc1-lock3-2-0-cust367.6-1.cable.virginmedia.com) |
| 12:58.32 | leifmadsen | we'll be showing how you modify C code in order to write dialplan |
| 12:58.57 | bip | but a GUI is still a necessity sometimes, I lobbied for havinour windows based pbx replaced by asterisk, now I m supposed to show em that is easier to configure and mantain ;-) |
| 12:58.58 | leifmadsen | </joke> |
| 12:59.06 | leifmadsen | bip: I didn't say it wasn't :) |
| 12:59.10 | bip | well I can just write hello world in c ... |
| 12:59.23 | leifmadsen | bip: learn how to do what you need in the GUI, then document it for your other admins |
| 12:59.27 | bip | so I guess I will skip that part ;-) |
| 12:59.29 | leifmadsen | that's basically the answer to your question |
| 12:59.31 | russellb | the GUI isn't too hard ... your first mistake was trying to mess with the files before you used the GUI |
| 12:59.39 | russellb | you should only use the GUI if you're going to use it |
| 12:59.43 | leifmadsen | ya never touch the files prior to the GUI |
| 12:59.50 | leifmadsen | let the GUI do all the work |
| 12:59.56 | bip | well i was tryin to reverse engineer the GUI |
| 12:59.56 | russellb | even if technically you can touch some of it ... just don't |
| 12:59.58 | *** join/#asterisk schmrz (~schmrz@usaidgbep1-178.bih.net.ba) |
| 12:59.59 | leifmadsen | otherwise it gets confused (which is reasonable to assume) |
| 13:00.15 | bip | if the GUI would read my hand edited file |
| 13:01.02 | schmrz | Hello Everyone! (again). Is there an known bug with ExtensionState returning 1 while the extension is actually `Idle` (0). |
| 13:01.05 | schmrz | *? |
| 13:01.09 | bip | yeah this what I hated about FreePBX |
| 13:01.26 | schmrz | It sometimes happens after hanging-up the phone. |
| 13:01.53 | bip | it forced me to use it ... now I wonder why command line editing and web based interface cannot just get alog sometimes :( |
| 13:01.56 | GhOnDiE | if i include a shell script in a dial plan ie exten => 4500,n,System(/var/lib/agi-bin/led_on.sh) |
| 13:02.11 | GhOnDiE | will that be stopped when the user answers the phone |
| 13:02.14 | GhOnDiE | ? |
| 13:02.26 | leifmadsen | russellb: you left me!? |
| 13:02.32 | russellb | leifmadsen: my bad |
| 13:03.53 | bip | what about things like asteriskNOW, trixbox, elastix does an easy to deploy asterisk exist ? or is just blood sweat and tears until it works ? |
| 13:04.11 | leifmadsen | all of those are easy to deploy -- it's the management through the GUI which is the trick |
| 13:04.18 | leifmadsen | don't modify the dialplan manually |
| 13:04.29 | leifmadsen | everything must be built from the GUI. It's an all or none scenario |
| 13:04.38 | russellb | if you want to avoid trouble anyway |
| 13:04.41 | bip | ok ... |
| 13:04.50 | bip | well I m already in trouble |
| 13:04.51 | leifmadsen | if you need custom dialplan and things custom to your environment, you either need to modify the GUI to understand what you're doing, or you need to build your own GUI |
| 13:04.57 | russellb | bip: start over :-) |
| 13:05.21 | bip | i have a digium card shipped next week, and my boss wants a pbx up and running :( |
| 13:05.21 | leifmadsen | like I said, the solution to your problem is documentation of how to modify things for your environment, or to build your own GUI (or as russell said, to start over) |
| 13:05.37 | bip | yeah I appreciate ... |
| 13:05.53 | leifmadsen | "Poor planning on your part does not constitute an emergency on mine." |
| 13:06.25 | bip | lemme ask a last question, I have foun a very simple tutorial that shows how to have a minimal config working from the command line: |
| 13:06.47 | bip | http://www.beardy.se/an-introduction-to-asterisk-the-open-source-telephony-project |
| 13:07.10 | bip | do you know anything that shows how to do the same with a GUI ? |
| 13:07.19 | florz | oops |
| 13:07.57 | *** join/#asterisk anonymouz666 (~anonymouz@189.25.8.72) |
| 13:07.57 | bip | because I had no problem following that and even tweakin it ... |
| 13:08.07 | zamba | Asterisk 1.6.2.5-0ubuntu1 - is this very outdated or still ok to use? |
| 13:08.26 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 13:09.02 | *** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca) |
| 13:12.00 | leifmadsen | well 1.6.2.13 exists |
| 13:12.07 | leifmadsen | and we release on a monthly cycle.... |
| 13:12.15 | leifmadsen | so 8 months out of date at least |
| 13:12.22 | *** join/#asterisk sezuan (bouncer@irc.scheff32.de) |
| 13:13.31 | anonymouz666 | leifmadsen as always the first one to use the latest asterisk on production |
| 13:13.34 | anonymouz666 | :P |
| 13:13.51 | leifmadsen | true :) |
| 13:13.59 | leifmadsen | I was using 1.4 in production when it was 1.4.0 :) |
| 13:14.06 | leifmadsen | I built a clustered ITSP out of it |
| 13:14.32 | leifmadsen | I've been running 1.8.0 since RC1 (as soon as the commercial modules were updated for 1.8) |
| 13:14.51 | anonymouz666 | running fine so far? |
| 13:15.08 | leifmadsen | yeppers |
| 13:15.20 | leifmadsen | use it for a conference call every day and all my peers are registered fine |
| 13:15.28 | leifmadsen | I found a but earlier, but that got fixed for 1.8.0 |
| 13:15.32 | leifmadsen | s/but/bug/ |
| 13:15.35 | *** part/#asterisk schmrz (~schmrz@usaidgbep1-178.bih.net.ba) |
| 13:16.15 | anonymouz666 | I'll wait a lit bit to move from 1.4 directly to 1.8. |
| 13:17.52 | schmidts | anonymouz666 it looks like i will upgrade from 1.2 to 1.8 ;) |
| 13:18.12 | drmessano | schmidts, oh, so you're the guy |
| 13:18.14 | zamba | but what is the best solution for us ubuntu users to still get updated asterisk? |
| 13:18.34 | drmessano | zamba, it's never been "use packages" |
| 13:18.35 | leifmadsen | zamba: use the source |
| 13:18.40 | drmessano | Source, source, source |
| 13:18.55 | leifmadsen | digium provides updated packages, but only RPMs |
| 13:19.28 | drmessano | leifmadsen, unless Qwell got drunk the night before |
| 13:19.54 | zamba | upgrading the source is a pain in the ass |
| 13:19.55 | drmessano | We all know he can be a slacker |
| 13:20.25 | drmessano | svn update && ./configure && make && make install never bothered me |
| 13:22.11 | WIMPy | nik_: Did you edit that log? It looks like something is missing around line 1877. |
| 13:22.16 | tzanger | slackware owns. |
| 13:23.50 | *** join/#asterisk rrb3942 (~rbullock@208.34.105.161) |
| 13:24.04 | *** join/#asterisk guillaumeb (~gbour@office-puteaux.avencall.com) |
| 13:24.12 | WIMPy | nik_: I don't see any attempt to accept that call this time, either. |
| 13:25.25 | [TK]D-Fender | [09:19]<zamba>upgrading the source is a pain in the ass <- Welcome to "life". |
| 13:26.09 | drmessano | Troubleshooting issues on 8 month releases is a pain in the ass too :) |
| 13:26.52 | *** join/#asterisk theHub (~karl@69.177.93.21) |
| 13:27.31 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:27.35 | *** join/#asterisk Dovid (Dovid@213.8.121.90) |
| 13:27.44 | WIMPy | nik_: All I can say is that the line state seems to get out of sync somehow. What versions are you running? Esp. of libpri? |
| 13:30.25 | drmessano | HAPPY INTERNATIONAL CAPS LOCK DAY |
| 13:33.17 | leifmadsen | HOLY EFF IS IT REALLY?! |
| 13:33.26 | leifmadsen | WE USUALLY HAVE CAPS LOCK FRIDAY |
| 13:33.58 | WIMPy | SAVE THE SHIFT, USE CAPS LOCK! |
| 13:34.31 | drmessano | http://thenextweb.com/socialmedia/2010/10/22/today-is-international-caps-lock-day-but-what-is-it |
| 13:35.56 | drmessano | HAHAH --> I CAN'T LOG IN TO GMAIL BECAUSE IT'S REJECTING MY PASSWORD. BECAUSE I'M NOT TURNING OFF CAPS LOCK. I STAND MY GROUND. #CAPSLOCKDAY |
| 13:36.05 | leifmadsen | lol |
| 13:37.08 | nik_ | WIMPy: did not edit logs |
| 13:38.25 | drmessano | My contribution: YAY, I GET TO SPEND ALL DAY TYPING THE WAY SAMUEL L. JACKSON SPEAKS!! #CAPSLOCKDAY |
| 13:38.27 | nik_ | WMIPy: libpri-1.4.11.4 |
| 13:38.55 | nik_ | WMIPy: libpri-1.4.11.4 & Asterisk 1.6.2.13 |
| 13:40.00 | WIMPy | nik_: Doesn't sound too bad. But unfortunaletly I don't reallt have an idea whet causes this. |
| 13:41.00 | nik_ | WIMPy: tried to search for "cause 81" on google, there are 6 entries in the world :( |
| 13:41.22 | nik_ | WIMPy: I was told just now that also long calls tend to get dropped |
| 13:41.39 | nik_ | WIMPy: calls longer than 3 minutes or so |
| 13:41.48 | WIMPy | nik_: Well, you should search for cause 51. Cause codes are usually written in hex. (and for a good reason) |
| 13:42.14 | WIMPy | What _exactely_ are you connected to? |
| 13:42.16 | nik_ | WIMPy: I was searching for the exact text as it appears in the logs |
| 13:42.35 | nik_ | 2 x ISDN2a lines |
| 13:42.41 | nik_ | 2 channels per line |
| 13:42.51 | nik_ | lights on the card are green |
| 13:42.58 | WIMPy | What hardware? |
| 13:43.07 | nik_ | lines appear to work normal via analogue ports |
| 13:43.07 | WIMPy | From the telco. |
| 13:43.31 | WIMPy | What analogue ports? |
| 13:43.33 | nik_ | digium B410P |
| 13:43.53 | nik_ | the ISDN2a is a box with 2 x analogue and 2 x digital ports |
| 13:44.07 | nik_ | so it is compatible with standard analogie phones |
| 13:44.17 | krion | hum |
| 13:44.32 | krion | i guess i've a problem with my dialplan |
| 13:44.32 | nik_ | we disconnected these since we connected the PBX though |
| 13:44.51 | WIMPy | nik_: Ok, so it's something active that could play part in the trouble. |
| 13:45.03 | krion | if someone as an emergency number as a short number, the emergency number is called instead |
| 13:45.14 | nik_ | WIMPy: no, nothing active to my knowledge |
| 13:45.43 | WIMPy | nik_: If it has analogue ports there has to be something active. |
| 13:45.49 | a1fa | is thinking about in-sourcing *... i just may need a beefy box to run all these great things on one virtual machine to stop burning so much power ;) |
| 13:47.28 | nik_ | WIMPy: not sure what you mean, what could be active? |
| 13:47.46 | nik_ | WIMPy: analogue phones are not connected |
| 13:48.00 | WIMPy | nik_: Your NT is obviousely more than just a NT. |
| 13:48.01 | nik_ | WIMPy: there are just analogue ports, unused |
| 13:49.15 | nik_ | WIMPy: yes, but what are you suggesting? it cannot be used with asterisk because of interference with the analogue ports? |
| 13:50.53 | WIMPy | nik_: No, I'm just saying that I don't know whats going wrong, but that box could also play a part in that. I needs not be entirely Asterisks fault. |
| 13:53.19 | *** join/#asterisk frigidzephyr (~rnewton@nat/digium/x-qeftvngdfpuovtbz) |
| 13:53.22 | nik_ | WIMPy: another log just captured: http://pastebin.com/FQ2bqNkk |
| 13:53.36 | *** join/#asterisk oryxtec (~aa@119.152.222.246) |
| 13:54.00 | oryxtec | guys has any one tested asterisk for production on virtual machine? |
| 13:54.55 | [TK]D-Fender | oryxtec: Hundreds of people no doubt |
| 13:55.30 | nik_ | WIMPy: any suggestions how do I go about it? |
| 13:56.09 | oryxtec | [TK]D-Fender: how is the result.. are there any issues or probelm using asterisk on VM |
| 13:56.10 | nik_ | WIMPy: I will have to work on this over the weekend, really need to find a solution or else will have to disconnect the PBX and use analogue phones without ability to transfer etc |
| 13:56.59 | nik_ | WIMPy: I am not sure how to approach the problem, where do I start? it is intermittent to begin with, so it is not trivial misconfiguation of something. |
| 13:57.12 | [TK]D-Fender | oryxtec: Potential timing issues for conferencing. this may vary based on load, VM type, etc. |
| 13:57.37 | [TK]D-Fender | oryxtec: What are your NEEDS and what are the specs fo the server you're looking at putting it on and what is the load of the other VM's on it? |
| 13:58.16 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 13:58.26 | oryxtec | i don't need much on VM.. i just wana install asterisk and give it one clint.. |
| 13:58.46 | oryxtec | he wont be doing any |
| 13:58.49 | [TK]D-Fender | oryxtec: huh? |
| 13:58.55 | oryxtec | confernace |
| 13:58.58 | WIMPy | nik_: As you've got Digium hardware, you could try their support. You could also experiment with other versions of libpri and/or dahdi. But I don't have andy idea what's going on there. |
| 13:58.59 | oryxtec | staff |
| 13:59.20 | [TK]D-Fender | heads off to a meeting... |
| 13:59.48 | anonymouz666 | [TK]D-Fender: don't forget to ask for a SIP debug there |
| 13:59.49 | anonymouz666 | :P |
| 14:04.53 | *** join/#asterisk msaraiva (~msaraiva@201.18.83.202) |
| 14:06.41 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 14:07.30 | *** join/#asterisk shapr (~shapr@nat/digium/x-xnnowwdlncwgxwnn) |
| 14:09.15 | *** join/#asterisk dongs (1000@l212168.ppp.asahi-net.or.jp) |
| 14:09.21 | dongs | so uh.. wat happen to skype for asterisk |
| 14:09.44 | dongs | i see a huge blog post on digium.com/skype which sends me to the "buy now" link at the store which tells me its a disabled category. |
| 14:09.53 | dongs | http://store.digium.com/productview.php?product_code=1SFA001 |
| 14:10.22 | shapr | dongs: Where's the blog post? |
| 14:10.26 | *** join/#asterisk rgsteele (~rgsteele@apo.aweber.com) |
| 14:10.32 | dongs | digium.com/skype << |
| 14:10.41 | dongs | and wehn you goto bulletpoint #18 |
| 14:10.44 | dongs | 'hwo do i buy it' |
| 14:10.57 | espiceland | http://store.digium.com/productview.php?product_code=1SFA0001 |
| 14:10.58 | dongs | If you are an end user, you can buy it from a reseller, or directly from the digium store. < the 'digium store' link goes to disabled category. |
| 14:11.00 | espiceland | The URL is wrong. |
| 14:11.21 | dongs | well, perhaps they want to fix it, if they want people to buy it. |
| 14:11.23 | drmessano | http://store.digium.com/productview.php?product_code=1SFA0001 |
| 14:11.28 | espiceland | Link to blog post? |
| 14:11.37 | drmessano | dongs, or you can go to store.digium.com and LQQK for it |
| 14:11.48 | dongs | drmessano: i was just typing that. try it. search for skype at the store. |
| 14:11.59 | nik_ | WIMPy: Thanks, I will try this, hope they are able to help with this |
| 14:12.07 | dongs | espiceland: http://www.digium.com/skype |
| 14:12.21 | espiceland | the "Buy Now!" button works fine for me |
| 14:12.26 | espiceland | Correct URL |
| 14:12.31 | dongs | which redirects to http://www.digium.com/en/products/software/skypeforasterisk.php |
| 14:12.37 | WIMPy | nik_: Or try again here later. Maybe someone else has an idea. |
| 14:12.37 | shapr | dongs: If I click the "Buy Now" button it works for me. |
| 14:12.49 | dongs | k, theurl in http://www.digium.com/en/docs/SFA/sfa_faq.php doesnt |
| 14:12.56 | dongs | under 18. how do i purchas |
| 14:13.20 | espiceland | *that* would be the link we were looking for. :) Thanks. |
| 14:13.41 | dongs | does this stuff still require a bunch of X11 libs |
| 14:13.50 | dongs | and runs skype client in background |
| 14:13.54 | dongs | or is this an actual proper solution |
| 14:14.17 | leifmadsen | dongs: that solution you speak of has nothing to do with the digium implementatin |
| 14:14.21 | shapr | dongs: What do you consider a proper solution? |
| 14:14.24 | leifmadsen | dongs: the digium implementation is a "real" driver |
| 14:14.36 | frigidzephyr | it does not require any x11 stuff |
| 14:14.38 | leifmadsen | dongs: native to asterisk with no external hacks -- if you want skype, this is the solution |
| 14:14.55 | leifmadsen | it's the only native skype channel driver -- the rest are hacks |
| 14:15.01 | dongs | leifmadsen: right, so it doesnt need X11 and doesnt run skype in background, and communicates to skypestuff natively / without hacks? |
| 14:15.04 | dongs | k |
| 14:15.08 | leifmadsen | that's what I just said |
| 14:15.13 | dongs | buying. |
| 14:15.21 | leifmadsen | uses it on his system |
| 14:15.23 | leifmadsen | works great |
| 14:16.39 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cpe-0210-4bff-fe2b-9074.dhcp.a2infotech.com) |
| 14:18.00 | dongs | geez why the fk does everthing require login/password now |
| 14:18.08 | dongs | and of course my login from 2000 doesnt work. |
| 14:20.58 | *** join/#asterisk evangelion (~manzy_zet@212.183.172.126) |
| 14:21.08 | evangelion | hello |
| 14:21.21 | evangelion | anyone who had problems with CISCO IOS and Asterisk around? we can't place calls from a Cisco2851 to an asterisk box |
| 14:21.36 | evangelion | i think that asterisk can't parse the invite requests that come from the router |
| 14:22.23 | evangelion | we tested with both auth and unauth requests |
| 14:22.33 | leifmadsen | evangelion: you're going to have to provide some debug information |
| 14:22.36 | angryuser | evangelion, i think you need to trace it actually before guessing |
| 14:24.51 | evangelion | leifmadsen angryuser: http://pastebin.com/ZfuAFv4N |
| 14:24.56 | evangelion | this is the request packet |
| 14:25.33 | leifmadsen | a single packet makes not a debug |
| 14:25.52 | leifmadsen | I can't see what asterisk is doing on the console, I don't see the responses from the other end or what asterisk is sending |
| 14:25.53 | *** join/#asterisk Faustov (~fst@gentoo/user/faustov) |
| 14:26.54 | evangelion | leifmadsen: be patient, please| http://pastebin.com/SJbf1Sdp |
| 14:27.08 | leifmadsen | I'm just telling you what I need |
| 14:27.18 | frigidzephyr | lol |
| 14:27.39 | leifmadsen | ok... so we have 2 packets now... |
| 14:27.40 | evangelion | leifmadsen: ack |
| 14:27.47 | evangelion | leifmadsen: strange thing is |
| 14:28.04 | evangelion | peer 1000 is configured to accept alaw,ulaw,g729 |
| 14:28.19 | dongs | evangelion: you guys are the reason we're running out of IPV4 address space. why doyou have all your phones on public internet addressess?? |
| 14:28.24 | evangelion | while asterisk reports Capabilities: us - 0xc (ulaw|alaw) |
| 14:28.25 | schmidts | evangelion but your client sent only g729 so its no wonder you will get a not accaptable here |
| 14:28.33 | suprstar | span 1 got hangup, cause 6 - anyone know what 'cause 6' means, or where I can find out? |
| 14:28.36 | leifmadsen | evangelion: do you have a codec_g729a.so file? do you have a license to use g729? |
| 14:28.41 | evangelion | ulaw,alaw are specified under [general] |
| 14:28.48 | tzanger | I've got a license to be awesome. |
| 14:28.56 | leifmadsen | one side speaks alaw and ulaw, the other side speaks g729 -- they can't communicate |
| 14:28.59 | leifmadsen | simple as that |
| 14:29.02 | p3nguin | You surely don't need both ulaw and alaw at the same time. |
| 14:29.04 | leifmadsen | changetype: modify |
| 14:29.05 | leifmadsen | add: AstAccountUserAgent |
| 14:29.05 | leifmadsen | AstAccountUserAgent: test |
| 14:29.05 | WIMPy | {0x06, "Channel unacceptable"}, |
| 14:29.09 | dongs | tzanger: rly |
| 14:29.09 | leifmadsen | bah |
| 14:29.10 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 14:29.12 | leifmadsen | Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - () |
| 14:29.20 | evangelion | leifmadsen: we have |
| 14:29.33 | p3nguin | combined = null |
| 14:29.35 | WIMPy | suprstar: What kind of interface? |
| 14:29.42 | leifmadsen | something is still wrong either with the configuration or the modules loaded |
| 14:29.52 | leifmadsen | it is matching on peer 1000 |
| 14:29.58 | leifmadsen | what does peer 1000 look like in sip.conf? |
| 14:30.15 | shapr | suprstar: Is your span 1 ISDN or what? |
| 14:30.20 | suprstar | WIMPy - e1's, diguim pe420 or some such... pcie 4-port card |
| 14:30.28 | tzanger | dongs: YARLY |
| 14:30.29 | schmidts | suprstar have a look at http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
| 14:30.41 | WIMPy | suprstar: And it's a full PRI? |
| 14:30.47 | suprstar | yes |
| 14:31.04 | evangelion | leifmadsen: http://pastebin.com/jeuRxnQ8 |
| 14:31.20 | evangelion | <leifmadsen> one side speaks alaw and ulaw, the other side speaks g729 -- they can't communicate <- ack! |
| 14:31.25 | WIMPy | suprstar: Then maybe you have one of these ancient configurations, where not all channels are allowed for both directions? |
| 14:31.31 | schmidts | evangelion: disallow=all and allow=g729 |
| 14:31.38 | leifmadsen | schmidts: he has that |
| 14:31.48 | leifmadsen | so from what I can tell the peer is g729 only and the other side doesn't speak g729 |
| 14:31.53 | leifmadsen | disallow=all |
| 14:31.54 | evangelion | leifmadsen: asterisk ignores the account part |
| 14:31.54 | leifmadsen | allow=g729 |
| 14:31.57 | leifmadsen | allow=ulaw |
| 14:32.11 | leifmadsen | evangelion: your statement doesn't parse |
| 14:32.13 | suprstar | WIMPy: afaik I'm not explicitly dis-allowing any channels |
| 14:32.27 | leifmadsen | evangelion: for that peer what is in the [...] |
| 14:32.34 | leifmadsen | I don't know what peer this is |
| 14:32.44 | WIMPy | suprstar: I'm talking about your telcos configuration. You can't configute such a thing. |
| 14:32.47 | suprstar | 'dahdi show channels' shows 93 channels ready for use |
| 14:32.49 | evangelion | leifmadsen: please check this: http://pastebin.com/SJbf1Sdp |
| 14:33.07 | leifmadsen | I saw that the first time |
| 14:33.21 | suprstar | WIMPy: could be, my telco is a corporate avaya phone switch. config prob there? |
| 14:33.22 | leifmadsen | the sides still only setup ulaw|alaw on one side, g729 on the other |
| 14:33.25 | evangelion | leifmadsen: asterisk reports Capabilities: us - 0xc (ulaw|alaw), |
| 14:33.27 | WIMPy | suprstar: 3 PRIs? Or what is it? |
| 14:33.41 | evangelion | leifmadsen: anyway the extension reports g729 only |
| 14:33.42 | leifmadsen | shrugs |
| 14:33.51 | suprstar | WIMPy: yes, 3 pri's on this server |
| 14:34.18 | leifmadsen | ok this is distracting, I need to be working on my own stuff |
| 14:34.20 | *** part/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:34.36 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:34.39 | WIMPy | suprstar: Can't tell you about the Avaya, but you could try to dial out different channels explicitely and see what happens on which channel(s). |
| 14:34.51 | WIMPy | suprstar: Are they all connected to the Avaya? |
| 14:35.33 | suprstar | WIMPy: yes, all connected to avaya. I can try calling out on diff. channels, you think one will end up non-usable? |
| 14:35.49 | WIMPy | suprstar: I guess so. |
| 14:36.25 | WIMPy | suprstar: Could also be a mismatched configuration. Three different lines vs a trunk. See -> NFAS. |
| 14:36.52 | suprstar | WIMPy: ok thx for the info, I'll check that out |
| 14:39.34 | BlackMaria | Anyone here familiar with ldap asterisk integration ( on debian ) ? |
| 14:47.45 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
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| 14:49.25 | *** join/#asterisk TJ_stamm (~TJ@69.197.107.254) |
| 14:55.52 | shapr | BlackMaria: I'm not, but what sort of questions do you have? |
| 14:59.17 | *** join/#asterisk rene- (~rene@189.202.122.239.cable.dyn.cableonline.com.mx) |
| 15:01.01 | rene- | hey guys, is it possible to get the durations of a call that was reinvited=yes (Native bridging) in the cdr? |
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| 15:29.29 | BlackMaria | skapr most of the examples dont work ( ldap schema issues ) |
| 15:30.26 | BlackMaria | ...but also I have the module res_config_ldap in the modules dir and I get an error about 'ldap' |
| 15:31.10 | BlackMaria | <PROTECTED> |
| 15:34.34 | Katty | HHHHHHHHHEEEEEEEEEEEEELLLLLLLOOOOOOOOOOOOOO Nurse. |
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| 16:11.40 | kalimc | Hello. I was wondering if anyone knows a good "DID" service, that will allow me to order numbers and point them to our gateway? |
| 16:11.51 | *** join/#asterisk UQlev (~yuriy@212.50.99.8) |
| 16:12.15 | ManxPower | ~itsp |
| 16:12.15 | 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. |
| 16:13.25 | kalimc | Any recommendations? |
| 16:13.44 | ManxPower | they all suck compared to PSTN |
| 16:13.50 | ManxPower | I personally use Vitelity.net |
| 16:15.57 | *** join/#asterisk chuckp (~chuckp@74-93-168-241-Jacksonville.hfc.comcastbusiness.net) |
| 16:16.09 | [TK]D-Fender | kalimc: Vitelity is fairly decent. Other good options : les.net , voip.ms |
| 16:16.21 | [TK]D-Fender | kalimc: Shop around |
| 16:16.34 | kalimc | ok, great, thanks guys. |
| 16:17.14 | *** part/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
| 16:17.49 | chuckp | For some reasons users that use the speakerphone on their Aastra 57i units tend to get strange popping and hissing sounds when they are not talking on a call and just listening. Anyone ever experience this? It seems to also occur on the Polycom SoundStations, which makes me think it is a server side setting |
| 16:17.57 | chuckp | I turned off Silence Supression |
| 16:19.32 | [TK]D-Fender | chuckp: And you aren't telling us clearly what is on the OTHER SIDE of that call. |
| 16:20.22 | *** join/#asterisk psilikon (~joel@cerberus.vicimarketing.com) |
| 16:20.29 | ManxPower | We had a customer complain today that the Polycom microphone was TOO GOOD |
| 16:20.42 | Naikrovek | nice |
| 16:20.48 | Naikrovek | picked up stuff they didn't want picked up? |
| 16:20.53 | ManxPower | yup. |
| 16:21.02 | shapr | chuckp: comfort noise? |
| 16:21.05 | ManxPower | I took 6dB off their Mic gain and they seem to be happy. |
| 16:21.08 | [TK]D-Fender | I got the with plantronic voice-tube based headsets. |
| 16:21.13 | [TK]D-Fender | Psycho sensitive |
| 16:21.15 | Naikrovek | there is also a mute button |
| 16:21.22 | *** part/#asterisk nik_ (c636cac3@gateway/web/freenode/ip.198.54.202.195) |
| 16:21.28 | [TK]D-Fender | ManxPower: wow, 6 db is a lot |
| 16:21.35 | ManxPower | shapr, if you don't have silence suppression then you do not have confort noise. |
| 16:21.38 | Naikrovek | 75% loss I think. |
| 16:21.46 | ManxPower | [TK]D-Fender, it was one of a long string of stupid complaints |
| 16:21.48 | Qwell | ManxPower: just go to lpc10 |
| 16:21.55 | Naikrovek | we had this conversation once and now i've forgotten what halfing the audio is in db |
| 16:21.55 | [TK]D-Fender | DOMO ARIGATO! |
| 16:22.04 | ManxPower | Qwell, do you have a Polycom firmware that supports that? |
| 16:22.15 | Qwell | ManxPower: transcode in Asterisk |
| 16:22.25 | ManxPower | oh, we don't do any transcodind |
| 16:23.26 | anonymouz666 | the codec I wanna see devices supporting is the speex but I haven't seen any |
| 16:28.29 | shapr | Qwell: oh hai |
| 16:28.39 | Qwell | shapr: wasn't me |
| 16:28.43 | Qwell | file did it |
| 16:28.56 | file | it's true |
| 16:29.23 | shapr | I don't believe it. |
| 16:29.25 | p3nguin | naikrovek: If you increase it 3 dB, you double it... so go from there. |
| 16:29.35 | Naikrovek | that's what i thought |
| 16:29.40 | Naikrovek | actually i thought it was 3.5db |
| 16:29.43 | Naikrovek | but whatever |
| 16:29.44 | Naikrovek | it matters not |
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| 16:43.50 | drmessano | Naikrovek, 3db is a 2x increase. I always remember telling the CB'ers that 4x the power was 6db, which happens to be 1 S-Unit on the signal meter of a CB radio. You tell them they need to quadruple their power to move someones meter 1 S-Unit and they look at you funny |
| 16:44.24 | Naikrovek | heh |
| 16:44.26 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 16:44.27 | Naikrovek | okay |
| 16:44.54 | [TK]D-Fender | Speaker power is 6db for double. |
| 16:44.55 | drmessano | "I modified this Wal Mart radio for 6 watts out!" "Umm, that's far from one S-Unit difference man" .. then they shoot you |
| 16:45.06 | psilikon | Anyone know of a supplier for used sangoma a108's besides ebay? |
| 16:45.14 | [TK]D-Fender | There is a difference depending on what kind of power you are comparing |
| 16:45.43 | [TK]D-Fender | psilikon: "ebay" isn't a supplier |
| 16:46.05 | *** join/#asterisk mintos (~mvaliyav@114.143.165.131) |
| 16:46.23 | psilikon | great |
| 16:47.14 | *** join/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
| 16:47.29 | kalimc | is 1.8 stable? Is it compatible with Fax For Asterisk? |
| 16:47.55 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
| 16:48.11 | [TK]D-Fender | kalimc: Stable enough for release. Stable enough for YOU is another matter. |
| 16:48.27 | [TK]D-Fender | kalimc: You you switch to a new version of your operating system the day it comes out? |
| 16:48.35 | coppice | [TK]D-Fender: 3dB is a doubling of power |
| 16:48.53 | kalimc | I agree with you, I'll use 1.6x for now. |
| 16:49.05 | [TK]D-Fender | coppice: I believe what I was referring to ont he speaker side was "perceieved invcrease" |
| 16:49.35 | [TK]D-Fender | kalimc: I would tend to wait until the first bug-fix release comes out... |
| 16:49.54 | kalimc | [TK]D-Fender - I took your advice with Vitelity, they are very in touch with asterisk. |
| 16:50.13 | coppice | [TK]D-Fender: while received increase is certainly logarithmic, can you say that something sounds twice as loud as something else? |
| 16:50.14 | kalimc | just installing asterisk on my new server now |
| 16:50.47 | coppice | [TK]D-Fender: 6dB is double the amplitude, which is 4 times the power |
| 16:51.11 | ManxPower | yeah, but what is that in terms of volume |
| 16:51.30 | coppice | several cubic metres |
| 16:51.38 | kalimc | Is DAHDI/LibPRI required if I plan only on SIP? (no pstn/pri's) |
| 16:51.47 | ManxPower | kaldemar, no |
| 16:52.09 | ManxPower | s/yeah, but what is that in terms of volume/yeah, but what is that in terms of perceived audio levels |
| 16:52.14 | [TK]D-Fender | SPL <- |
| 16:54.51 | coppice | 6dB is perceived as only a modest increase in volume. however, I have no idea what twice the volume sounds like. i'm pretty sure its a low more than 6dB |
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| 16:59.32 | *** join/#asterisk settntrenz (~joe@137.216.121.70.cfl.res.rr.com) |
| 17:00.20 | p3nguin | drmessano: What pisses me off is that if I'm already running nearly a kW, I have to run ANOTHER full kW to make any noticeable difference on the receiving end. |
| 17:01.04 | settntrenz | I've got an asterisk server that supports several satellite offices and we recently had some employees move into a new office that has a sonicwall pro firewall. The cisco IP phones in that office can make outgoing calls but not receive incoming calls. Their sip peers are configured with NAT=YES. On the Sonicwall, Enable Consistent NAT is checked, as is Enable SIP Transformations. A sip debug of the peer in question shows several restransmit |
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| 17:02.42 | Naikrovek | settntrenz: wow i think you work where i work. but maybe not |
| 17:02.49 | Naikrovek | i'm having that exact same issue |
| 17:03.10 | Naikrovek | s/maybe/definitely |
| 17:04.18 | kalimc | for sip, do I open UDP or TCP (or both) |
| 17:04.20 | settntrenz | Naikrovek: how sure of that are you? |
| 17:04.35 | Naikrovek | well, based on your hostname, really |
| 17:04.40 | p3nguin | kalimc: Most implementations don't use TCP for SIP. |
| 17:04.46 | kalimc | ok, thanks |
| 17:04.53 | Naikrovek | settntrenz: the only people in my company besides myself who would care about this problem are in india |
| 17:04.54 | p3nguin | Asterisk implementations, that is. |
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| 17:06.59 | drmessano | p3nguin, unless you clip the swing diode |
| 17:07.06 | Naikrovek | settntrenz: but it is the EXACT same issue i'm having |
| 17:07.33 | p3nguin | drmessano: I run a Kenwood, so I don't think it has a "swing diode" in it. |
| 17:07.35 | settntrenz | Naikrovek: well, I don't think we work together, but I'm all for working together to fix this |
| 17:07.47 | Naikrovek | and i told them not to get a sonicwall. they got it because they have experience using it, but guess who had to tell them how to use the thing |
| 17:07.58 | Naikrovek | settntrenz: your english is FAR too good to be anyone I work with |
| 17:07.59 | drmessano | p3nguin, I am not sure my TS-520s has a diode in it |
| 17:08.08 | coppice | I bought a Kenwood today. A vacuum cleaner |
| 17:08.22 | settntrenz | Naikrovek: lol, sounds about right. We don't own/operate the sonicwall, which makes it much more difficult to try to fix. |
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| 17:08.51 | anonymouz666 | coppice: it follows the spec? |
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| 17:09.14 | drmessano | coppice, I would love to send a week shopping in HK.. I would love to bring home a Sony toaster and a Toshiba toilet |
| 17:09.21 | drmessano | spend* |
| 17:09.22 | coppice | well, it really sucks |
| 17:09.24 | p3nguin | drmessano: I've got a 440... I've never had it open. I bought it already modded for full-range transmit (including 60m). |
| 17:09.25 | settntrenz | Naikrovek: It's one of those situations where several clients in the same building share an internet connection provided by the building management company. All the other clients in the building that use VOIP have working solutions, but ours is not. Despite working everywhere else BUT in that building. |
| 17:09.44 | coppice | Toshiba do make toilets |
| 17:10.01 | coppice | and I think Sony made things like toasters at one time |
| 17:10.02 | settntrenz | Naikrovek: Unfortunately all the google searches I've done are unsolved reports of the same issue. |
| 17:10.12 | drmessano | p3nguin, as soon as my Fiancee gets her ticket, we may buy a newer HF rig, but between all the mobile rigs, HT's, and the upkeep of my repeater, my radio budget gets blown quickly |
| 17:10.25 | Naikrovek | well if you set up a site-to-site vpn it'll work - the sonicwall won't mess with the vpn packets, but even when you turn off sip transformations the sonicwall will still try to "help" you |
| 17:11.14 | p3nguin | drmessano: I had to stop with the radio stuff when I moved into an apartment that doesn't allow for an antenna outside. Some day I'll get back into it, I'm sure. |
| 17:11.15 | settntrenz | Naikrovek: Yeah, I was afraid of that the sonicwall was introducing some type of double NAT |
| 17:11.32 | Naikrovek | yeah i haven't figured it out either, but it's doing something |
| 17:11.54 | p3nguin | drmessano: I've still got old CB gear, but I can't make myself mount an RCI-2950 in my Benz. |
| 17:12.28 | coppice | Kenwood most certainly make toasters |
| 17:12.49 | p3nguin | Actually, I could probably get the radio installed, but I don't want to mount a Wilson 1000 on the roof. |
| 17:12.51 | settntrenz | Naikrovek: I tried disabling NAT for the sip peer, that didn't work. |
| 17:13.27 | Naikrovek | for a time, i gave the phones public IP addresses, and even that didn't fix it. the sonic wall is up to some nefarious shit with sip |
| 17:14.09 | settntrenz | Naikrovek: I'm pretty sure it alters the packets and puts the IP/port that it thinks should be there. |
| 17:14.24 | Naikrovek | something like that |
| 17:14.38 | Naikrovek | settntrenz: they hate it when i'm right about stuff like this, too |
| 17:14.52 | settntrenz | Naikrovek: of course, just be sure to give em a good "told ya so" |
| 17:15.08 | Naikrovek | settntrenz: already done that. the other offices have Cisco ASAs and they work great |
| 17:15.41 | settntrenz | Naikrovek: our other offices have internet connections we actually manage. |
| 17:16.09 | Naikrovek | the building management company is spying on everyone in that building, i'm betting |
| 17:16.50 | settntrenz | yeah, overall it's a bad idea. Not only the building management but whats stopping other LAN users from arp poisoning and mitm. |
| 17:17.59 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 17:19.50 | adelas | anybody know how to turn off the hardware canceller on the digium cards? I recall it being a parameter for the /etc/asterisk/chan_dahdi.conf |
| 17:20.24 | Naikrovek | why would you |
| 17:20.42 | adelas | i can reproduce dropped calls |
| 17:20.51 | adelas | i have a feeling its the canceler card |
| 17:20.57 | adelas | b/c i dont' get dropped if i use the backup voip line |
| 17:21.05 | WIMPy | adelas: It's a module parameter. Try modinfo. |
| 17:21.15 | Naikrovek | there's a lot more different between those two than the echo cancellation |
| 17:22.19 | adelas | okay, so my situtation right now is, when i call gotomeeting for a conference call, i dial a pin conntaining 4480, when it reads back that specific numbers, it droped right before saying 0 |
| 17:22.44 | adelas | but if i enter a different set of pin #, it is able to read it back perfectly fine |
| 17:23.59 | adelas | with the TE220 card with hardware canceller, i have to enable "relaxdtmf=yes" in the config |
| 17:24.23 | adelas | to get incoming dtmf to detect correctly |
| 17:24.53 | adelas | before i was using a TE120P, and didn't have to use that, didn't have any issues, and didn't have noise canceller |
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| 17:43.46 | adelas | damnit i'm just gonna yank that cancceler off the card |
| 17:43.49 | adelas | can't figure out hte command |
| 17:43.59 | Naikrovek | i was going to suggest that |
| 17:44.09 | Naikrovek | if it is feasible to do that during business hours |
| 17:44.26 | adelas | can't, gonna do that later after hours |
| 17:44.28 | WIMPy | adelas: I told you: |
| 17:44.31 | WIMPy | adelas: It's a module parameter. Try modinfo. |
| 17:44.45 | Naikrovek | i'll be surprised it's the EC module but I'm wrong all the time |
| 17:45.28 | adelas | can you be more specific about the modinfo? i'm not familiar with that |
| 17:45.37 | settntrenz | Naikrovek: I'm gonna have the sonicwall admin change a few things; udp-connection-memory timeout >=60, and several variations with enabling/disabling the persistent NAT and Sip Transformations. I'll let you know what happens. |
| 17:45.47 | Naikrovek | please do |
| 17:45.48 | Naikrovek | thank you |
| 17:46.19 | WIMPy | adelas: modinfo wxt4xxp or whatever your driver is. That will list the parameters. |
| 17:47.57 | *** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright) |
| 17:49.43 | adelas | WIMPy: i did a modinfo wxt4xxp, and it listed me a bunch of alies: and parms |
| 17:49.54 | adelas | would you like to see? |
| 17:50.43 | Naikrovek | pastebin |
| 17:51.34 | adelas | http://pastebin.com/dK4nMt7n |
| 17:52.57 | WIMPy | adelas: What do you think about vpmsupport? |
| 17:53.30 | adelas | what does tha tmean? |
| 17:53.31 | WIMPy | Actually it looks like you can disable only HW DTMF detection with vpmdtmfsupport. |
| 17:53.57 | WIMPy | The module is valled vpm-something, in case you hadn;t noticed. |
| 17:54.34 | WIMPy | is off fot a nice video evening. |
| 17:55.12 | Katty | OHIO!! |
| 17:55.56 | adelas | not sure how you would disable that vis software |
| 17:56.48 | adelas | okay, i'll just yank the damn thing off later and see how it goes |
| 17:57.19 | adelas | hopefully it won't blow my system |
| 17:59.09 | Katty | hai? |
| 17:59.14 | Katty | pokes at people |
| 17:59.16 | shapr | oh hoi |
| 17:59.28 | Katty | hello my asterisk does not work at all how to fix pls????? |
| 17:59.46 | adelas | whats the issue? |
| 17:59.55 | Katty | it does not work. |
| 18:00.14 | adelas | it think you need to explain the setup |
| 18:00.18 | Katty | you are just so cute. |
| 18:00.22 | Katty | pats adelas |
| 18:00.44 | Katty | look at this guy, he actually thought i was serious. |
| 18:01.05 | Katty | that's adorable, really. |
| 18:01.06 | *** join/#asterisk mrfrenzy (~mrfrenzy@abydos.syntaxis.se) |
| 18:01.58 | adelas | umm maybe i should stick around this chatroom more often to spot out the comedians in teh chan |
| 18:02.21 | Katty | maybe you should! |
| 18:02.36 | adelas | but than again, i should be at work :( |
| 18:02.54 | Katty | that's what multitasking is for |
| 18:03.22 | Katty | i'm talking to you, and eating a cupcake. |
| 18:03.23 | adelas | i shuoldn't be caught vpning into my home computer to be on irc :D |
| 18:03.28 | Katty | which takes skills. |
| 18:03.58 | Katty | adelas: when you get off work, you should buy a box of twinkies and put them in the freezer. |
| 18:05.10 | adelas | will the explode? |
| 18:05.12 | adelas | they* |
| 18:05.14 | shapr | hej mrfrenzy |
| 18:05.22 | Katty | adelas: mmmm, no |
| 18:05.47 | mrfrenzy | hej shapr ;) do I know you? |
| 18:06.00 | Katty | adelas: frozen twinkies are awesome. |
| 18:06.04 | Katty | adelas: you should try it |
| 18:06.24 | Katty | especially after dinner. |
| 18:06.30 | mrfrenzy | I'm trying to get outcall(windows app to dial through asterisk manager interface) to work with pap2 |
| 18:06.39 | Katty | Naikrovek: and you are going to miss an awesome dinner. |
| 18:06.41 | cusco | adelas: error messages? |
| 18:06.45 | mrfrenzy | after asterisks talk to the pap2 it says sip 302 "Moved Temporarily" |
| 18:06.47 | Katty | Naikrovek: i'm making chili, with spaghetti |
| 18:06.50 | Katty | Naikrovek: and cornbread. |
| 18:06.51 | mrfrenzy | then asterisk just gives up the connection |
| 18:06.56 | Katty | Naikrovek: and fritos |
| 18:06.56 | adelas | for the drop call? |
| 18:07.01 | Naikrovek | yum |
| 18:07.13 | Katty | Naikrovek: and hot dogs. |
| 18:07.14 | cusco | adelas: for whatever reasonit doesn't work? |
| 18:07.26 | shapr | mrfrenzy: Nej, men, om du vill... |
| 18:07.31 | Katty | Naikrovek: so i'm pretty sure it should last all weekend in some variation |
| 18:07.34 | adelas | theres no error msg, but theres a debug messsage of the call being dropped, let me reproduce it and gvive to you |
| 18:07.55 | Katty | Naikrovek: were you a smidgen closer, i'd invite you over for dinner. |
| 18:07.56 | cusco | adelas: use pastebin |
| 18:08.01 | cusco | also paste dialplan |
| 18:08.12 | shapr | mrfrenzy: Is there much use of Asterisk in Sweden? |
| 18:08.24 | Naikrovek | Katty: i'd be honored but yeah it's a bit far |
| 18:08.33 | *** part/#asterisk Mhaddog (~Mhaddog@173-102-30-203.pools.spcsdns.net) |
| 18:10.00 | adelas | cusco: theres not much, but this is what happens right when i get d/c |
| 18:10.02 | adelas | http://pastebin.com/CuBjXMge |
| 18:10.40 | mrfrenzy | shapr: yes quite a lot, there is a small community here |
| 18:10.44 | adelas | i think it ssomething due to the first line |
| 18:10.44 | Katty | also! |
| 18:10.48 | Katty | Naikrovek: http://allrecipes.com/Recipe/Smooth-Sweet-Tea/Detail.aspx <- |
| 18:11.02 | mrfrenzy | http://paste.debian.net/97677/ if anyone has an idea what could be the problem I would be grateful |
| 18:11.02 | Katty | Naikrovek: baking soda...whodathunk it? |
| 18:11.10 | Naikrovek | not me |
| 18:11.38 | adelas | cusco: so the other person is reading back numbers, and it just gets dropped after saying 4488 dropp, the next number would be 0 |
| 18:12.38 | Katty | anywho i think it might be time for a nap |
| 18:12.48 | cusco | adelas: ast_dsp_busydetect |
| 18:12.54 | cusco | who is answering the call? |
| 18:12.58 | cusco | adelas: dialplan? |
| 18:13.02 | shapr | mrfrenzy: Nifty, any Swedish Asterisk forums? |
| 18:13.29 | mrfrenzy | shapr: http://www.voip-forum.se/ |
| 18:13.35 | adelas | cusco: what do you mean answering? I dialed the number, punched in 44880#, then the system plays back the number for confirmation |
| 18:14.03 | cusco | where is the dial on that paste? |
| 18:14.15 | adelas | oh way further up |
| 18:14.18 | mrfrenzy | there was a bigger forum but it is unfortunately dead |
| 18:14.18 | *** part/#asterisk settntrenz (~joe@137.216.121.70.cfl.res.rr.com) |
| 18:14.23 | adelas | i'm placing the dial |
| 18:14.27 | cusco | adelas: and where is 44880# ? |
| 18:14.35 | adelas | um its lines above |
| 18:14.42 | cusco | and dialplan? |
| 18:14.43 | shapr | mrfrenzy: vad kul, tack! |
| 18:14.50 | adelas | okay let me get you a bigger copy and paste |
| 18:14.57 | adelas | its flooded with other stuff |
| 18:15.44 | adelas | is it possible to filter out the thing that shows in console to only show my extension? |
| 18:16.15 | cusco | adelas: I normally grep the log for say... 21945 as shown on "DEBUG[21954]" |
| 18:17.35 | *** join/#asterisk dwery (~dwery@nslu2-linux/dwery) |
| 18:18.56 | *** join/#asterisk EiNSTeiN_ (~einstein@unaffiliated/einstein/x-615171) |
| 18:20.00 | dwery | Hello. I'm searching for betatesters for an asterisk related iPhone app. Please read http://goo.gl/2adw for details. Thanks |
| 18:23.27 | *** join/#asterisk Jasnejac (~kvirc@81.91.106.59) |
| 18:23.37 | adelas | cusco: heres a complete call http://pastebin.com/P4FV4M4a |
| 18:23.45 | adelas | i mean complete as from dial to drop |
| 18:25.37 | shapr | I would like better asterisk log analysis tools. |
| 18:28.29 | adelas | cusco: I really hope you can spot a problem |
| 18:28.33 | adelas | on why its being dropped |
| 18:33.02 | cusco | adelas: im not really an expert, I am only trying to make yourself express better, so perhaps you can be helped by someone |
| 18:33.24 | cusco | tho the problem seems to start right on the dial |
| 18:33.29 | cusco | line 106 of your paste |
| 18:34.00 | cusco | I don't know what is ast_dsp_set_busy_pattern |
| 18:34.52 | cusco | you cut some part out where asterisk reads back the numbers |
| 18:35.00 | cusco | line 186 on your paste.. |
| 18:35.19 | cusco | and this is why teh call is hanged up: [2010-10-22 11:20:23] DEBUG[22550]: dsp.c:1372 ast_dsp_busydetect: ast_dsp_busydetect detected busy, avgtone: 255, avgsilence 265 |
| 18:35.44 | cusco | but what happens just before, you cut out |
| 18:35.55 | adelas | i didn't cut any part out |
| 18:36.06 | adelas | i just inserted that comment |
| 18:36.09 | cusco | line 187 on your paste |
| 18:36.42 | adelas | from line 185 to 188 its just the other person reading back |
| 18:36.46 | adelas | theres no logs |
| 18:36.52 | adelas | its just voice read back |
| 18:36.57 | adelas | of the numbers i just entered |
| 18:37.07 | ANurmi | does anyone know the default account and password for a Polycom soundpoint ip430? |
| 18:37.17 | Naikrovek | Polycom/456 |
| 18:37.26 | ANurmi | thank you much |
| 18:37.27 | adelas | i just put some blank lines there so you can see when i got the drop call |
| 18:37.55 | *** join/#asterisk mpe (~mpe@0xd99d3f8f.customer.cybercity.dk) |
| 18:39.29 | adelas | so technically, line 185 and 188 is next to each other |
| 18:39.34 | adelas | nothing in between |
| 18:48.46 | ManxPower | cusco, Busy Detect seldom solves more problems than it creates. |
| 18:50.57 | ChannelZ | Random Disconnection Enforcement |
| 18:53.17 | *** join/#asterisk cRAT (~cRAT@jasper.afg.com) |
| 18:56.05 | *** join/#asterisk nitrus^ (~Scott@99-52-103-128.lightspeed.sndgca.sbcglobal.net) |
| 18:57.06 | nitrus^ | select count(*) from cdr where length(src) = 3 and length(dst) > 3; does this CDR query make sense to find the total count of outbound calls from internal extensions that are 3 digits long to non extensions? the query is only giving me back a total of 3, which makes absolutely no sense |
| 18:58.14 | nitrus^ | or will that only word if the CID on the phones are all set to their extensions? |
| 19:03.52 | *** part/#asterisk wizhippo (~Adium@64.201.57.7) |
| 19:05.38 | nitrus^ | looks like query for a channel that has my extension in it and a dst that's got a length >3 makes more sense |
| 19:15.17 | *** join/#asterisk a1fa (~a1fa@unaffiliated/a1fa) |
| 19:15.18 | ManxPower | change your count(*) to a * and see what calls are matching |
| 19:36.44 | *** join/#asterisk war9407 (war@liquidswords.org) |
| 19:37.32 | cusco | ManxPower: I believe you, I just never happened to use it... |
| 19:47.47 | *** join/#asterisk rocky3 (~rocky@184-15-122-210.dr02.chtn.wv.frontiernet.net) |
| 19:49.32 | *** join/#asterisk ferdna (~yup@cpe-24-92-114-97.elp.res.rr.com) |
| 19:56.53 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 20:06.28 | *** join/#asterisk bjornts (~Adium@247.62-97-195.bkkb.no) |
| 20:06.58 | *** join/#asterisk ritzt3ch (ritzt3ch@ip70-189-221-21.lv.lv.cox.net) |
| 20:12.59 | *** join/#asterisk Micc (~quassel@66.216.207.185) |
| 20:17.42 | ritzt3ch | aloha : ) |
| 20:18.16 | *** join/#asterisk bluOxigen (~sfds@unaffiliated/bluOxigen) |
| 20:18.20 | ritzt3ch | i think i found a bug in asterisk on why some of my calls wouldnt go through |
| 20:18.27 | ritzt3ch | i checked my /var/log/asterisk/full |
| 20:18.28 | ritzt3ch | Call failed to go through, reason (3) Remote end Ringing |
| 20:19.15 | *** join/#asterisk fofware (~Fabian@host244.186-110-118.telecom.net.ar) |
| 20:19.29 | p3nguin | ~collectdebug |
| 20:19.29 | infobot | collectdebug is probably a method of collecting logs allowing others help troubleshoot an issue. Refer to doc/HOWTO_collect_debug_information.txt or http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt |
| 20:19.32 | p3nguin | File an issue. |
| 20:19.54 | Qwell | It's highly unlikely that there is a bug causing your calls to fail. |
| 20:21.15 | ritzt3ch | well it all points to it haha .... im paging (when paging i dial 4 outbound) using 4 dahdi lines to page ... and almost always 1 of the 4 lines fail |
| 20:21.20 | ritzt3ch | and i saw that |
| 20:22.10 | ritzt3ch | and my tech had a buttset on the line and he will SOMETIMES hear a dialtone .... and sometimes not when he doesnt the call/page fails. |
| 20:22.20 | ritzt3ch | but the cli shows everything fine |
| 20:26.29 | ManxPower | ritzt3ch, do the calls come in starting at channel 1 and going up, or 4 and going down |
| 20:26.57 | ManxPower | or are these lines ONLY used for paging? |
| 20:27.50 | *** join/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51) |
| 20:29.42 | *** part/#asterisk c0rnoTa (~c0rnoTa@80.251.113.51) |
| 20:30.32 | ritzt3ch | i believe all 4 |
| 20:31.01 | ritzt3ch | 1 - 4 |
| 20:31.38 | *** join/#asterisk itiliti (~itiliti@74.222.60.226) |
| 20:31.41 | ritzt3ch | and they are only used for paging |
| 20:32.06 | itiliti | can I include content in a SIP notify? |
| 20:32.16 | itiliti | in Asteisk 1.6 or 1.8? |
| 20:33.27 | ManxPower | ritzt3ch, did you do something silly like enable busydetect or callprogress on those ports? |
| 20:36.45 | *** join/#asterisk leroybuckingham (637c543e@gateway/web/freenode/ip.99.124.84.62) |
| 20:38.06 | *** join/#asterisk kalimc (~mcurry@proxy.hostopia.com) |
| 20:38.32 | kalimc | I am going to get some recordings done by an actor, once I get them back, what format should I convert them to for 'PlayBack' ? |
| 20:39.05 | ManxPower | kaldemar, convert them to every codec you use. |
| 20:39.44 | mrfrenzy | many of those who provide recordings already deliver many formats |
| 20:39.50 | mrfrenzy | http://www.westany.com/lilly_female_swedish for example |
| 20:40.05 | mrfrenzy | We provide the entire set of recordings in alaw, ulaw, gsm and sln |
| 20:45.17 | ritzt3ch | i wanted to do callprogress but that didnt take |
| 20:48.28 | ManxPower | there is a reason it says in the sample config file not to use it |
| 20:48.34 | ritzt3ch | I didnt try busy detect but its off |
| 20:48.49 | ritzt3ch | ive tried many things |
| 20:48.57 | ritzt3ch | i have dahdi 2.4 |
| 20:49.02 | ritzt3ch | and asterisk 1.6 |
| 20:49.09 | ManxPower | did you try using a paging system with proper disconnect indication? |
| 20:51.56 | p3nguin | If I have a phone that has low audio, is there anything in Asterisk that can be done to raise the levels on it? |
| 20:52.06 | Qwell | func_volume |
| 20:52.40 | citywok | wow, i hadn't read anything about 1.8, it sounds pretty cool. routing based on calendar busy status is something i was asked for a long time ago. |
| 20:57.44 | *** join/#asterisk deltalyst (~haryv@154.5.144.132) |
| 20:59.36 | Katty | ATTENTION |
| 20:59.41 | *** join/#asterisk seanjohn (~seanjohn@gateways.sheltoncomputers.com) |
| 20:59.43 | Katty | if you had a welsh corgi puppy--what would you name it? |
| 21:00.26 | *** join/#asterisk aut (~aut@c-174-48-60-12.hsd1.fl.comcast.net) |
| 21:00.34 | seanjohn | shouldn't disallow=all and allow=ulaw&g729&... be "rtp_allow" and "rtp_disallow"? |
| 21:00.41 | aut | does anyone know how to reset the password on a soundpoint 330? the defualt 456 isn't working on one of my phones |
| 21:00.49 | Qwell | seanjohn: no |
| 21:01.06 | seanjohn | Qwell, i know thats not whats in asterisk |
| 21:01.35 | seanjohn | thats why i'm saying, since those only pertain to the codec, or rtp packets, they should be prefixed with rtp_ |
| 21:01.37 | Qwell | seanjohn: The same options are used in every channel driver. |
| 21:02.03 | seanjohn | oh, for iax and t38 |
| 21:02.24 | seanjohn | iax_code and t38_codec? |
| 21:02.29 | seanjohn | codec* |
| 21:02.37 | ManxPower | I doubt T.38 will work with IAX |
| 21:02.41 | Qwell | or we could call them "allow" and "disallow". |
| 21:02.45 | seanjohn | no, manx, keep up |
| 21:03.05 | seanjohn | allow is more for security |
| 21:03.10 | seanjohn | allow/disallow |
| 21:03.20 | ManxPower | <seanjohn> iax_code and t38_codec? |
| 21:03.31 | seanjohn | they could have even made them codec= |
| 21:03.37 | Qwell | It's open source. Change them on your system if you like. |
| 21:03.40 | seanjohn | codecs_allow= |
| 21:03.43 | Qwell | We aren't changing it, however. |
| 21:03.48 | ManxPower | as the sample config files say, allow/disallow is for codecs, permit/deny is for access control |
| 21:04.37 | seanjohn | i'm not asking you to change it qwell. I'm asking if you thought that would have been better. It would cause everyone to have to change their configurations and is not a major priority |
| 21:04.49 | Qwell | No, it wouldn't have been better. |
| 21:04.52 | seanjohn | so, no, it shouldn't be changed |
| 21:05.26 | *** join/#asterisk fofware (~Fabian@host25.186-110-117.telecom.net.ar) |
| 21:06.24 | Qwell | seanjohn: What is your point? |
| 21:07.53 | seanjohn | our termination and originations use asterisk for their direct connection to nanpa. Their config says to use ONE sip context "[whatever]" for both incoming and outgoing and use type=peer. However, they have a problem with the disposition status or sip event messages. Half the time the calls show still progressing while the call is connected and other times the line is busy, they send "answered". |
| 21:08.17 | *** part/#asterisk ups (~ups@74-93-178-213-Sacramento.hfc.comcastbusiness.net) |
| 21:09.06 | ManxPower | I don't see how that relates to cosmetic changes in the config file options? |
| 21:09.17 | seanjohn | it doesn't |
| 21:09.25 | titter | Then why mention it? |
| 21:09.34 | seanjohn | would you know why,Qwell, since you're a developer |
| 21:10.04 | seanjohn | they said they verified the messages from the level 3 carrier as being transmitted correctly |
| 21:10.17 | seanjohn | but their equipment wasn't transmitting it on correctly |
| 21:11.12 | seanjohn | i think they compiled it wrong |
| 21:11.16 | titter | Then skip the middle man and call Level 3 :p |
| 21:11.44 | seanjohn | these people are level3. They have their own NANPA registrations |
| 21:11.54 | seanjohn | from their mother company |
| 21:12.28 | seanjohn | their billing is in sweeden but their sip and iax points are in the United States and UK |
| 21:12.43 | seanjohn | a lot of GOOD, local 20 ms points |
| 21:13.12 | seanjohn | the call quality is awesome |
| 21:13.15 | titter | I use Level 3 as my PSTN directly. Never heard of nanpa, but who knows |
| 21:13.37 | seanjohn | pstn is limited to how many lines? 24? |
| 21:13.48 | titter | Whatever I want |
| 21:13.59 | titter | I am setup with SIP trunking with Level 3 |
| 21:14.43 | seanjohn | well they have pstn and I pay .07 PENNIES per minute in 6 second increments wholsale. I can handle 120 calls but only use 110 in factors |
| 21:14.56 | titter | I am getting .02/min |
| 21:14.59 | titter | .03 switched |
| 21:15.09 | seanjohn | how much do you pay for the pstn |
| 21:15.19 | seanjohn | the T1 |
| 21:15.22 | titter | Its 15.10 per call path I have 80 |
| 21:15.25 | titter | I don't have a T1 |
| 21:15.32 | titter | Level 3 provides me with a SIP trunk |
| 21:15.49 | citywok | is there any new documentation on the new exchange calendaring features? |
| 21:16.00 | titter | I also have a fail over in another data center if for some reason I lose one server |
| 21:16.47 | seanjohn | for the 4000 i average daily, 40 at any given time, I only pay 300 |
| 21:17.18 | titter | *shurgs* |
| 21:18.04 | citywok | seanjohn you only pay $300/mo for 4000 calls daily? how long are your calls? |
| 21:20.52 | seanjohn | the other day was 18 hours |
| 21:21.04 | seanjohn | yesterday 12 hours they all vary. this is residential using me |
| 21:23.51 | *** part/#asterisk GameGamer43 (~GameGamer@li119-60.members.linode.com) |
| 21:32.02 | citywok | i'm looking at calendar.conf and you have to give a username/password for exchange support. is it possible to use a single user/password that has access to ALL accounts, and have it access the right mailbox on the fly? this would remove the need to ahve every users domain password. |
| 21:37.16 | *** join/#asterisk DrkShadow (~andrew@host-72-175-240-62.static.bresnan.net) |
| 21:37.29 | DrkShadow | How can I add someone to a queue from... outside of asterisk? I want to do it with a shell script. |
| 21:37.39 | *** join/#asterisk csnook (~chris@209-6-38-66.c3-0.smr-ubr1.sbo-smr.ma.cable.rcn.com) |
| 21:43.22 | pabelanger | DrkShadow: *CLI> core show application AddQueueMember |
| 21:43.40 | DrkShadow | add queue member <member> to <queue> |
| 21:45.03 | DrkShadow | ... queue add.. |
| 21:45.33 | ritzt3ch | wait for dialtone HOW is that set in dahdi |
| 21:47.53 | DrkShadow | Can I set a global variable for all channels through the command line? |
| 21:48.08 | DrkShadow | core set global?.. |
| 21:48.12 | DrkShadow | I dunno how to test it.. |
| 21:49.44 | *** join/#asterisk nny (~Scott@174.107.201.103) |
| 21:49.49 | nny | holy wtf |
| 21:50.49 | nny | chan_sip.c: Unable to allocate AST channel structure for SIP channel |
| 21:52.33 | pabelanger | That's not good |
| 21:52.39 | nny | yeah had to reboot linux |
| 21:52.44 | nny | apparently hit a 10 month limit lol |
| 21:52.53 | nny | at about 1.5 million calls |
| 21:53.24 | DrkShadow | up 368 days, 23:34 |
| 21:53.25 | pabelanger | impressive |
| 21:53.33 | DrkShadow | dunno how to see how many calls |
| 21:53.43 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 21:55.42 | Qwell | DrkShadow: core show calls |
| 21:56.14 | DrkShadow | "No such command 'core show calls' |
| 21:56.22 | Qwell | You're using a sucky version then. |
| 21:56.31 | nny | wow |
| 21:56.32 | DrkShadow | though if I had to guess I'd say 100-200 000 |
| 21:56.33 | nny | http://lists.digium.com/pipermail/asterisk-users/2006-April/147204.html |
| 21:56.34 | Qwell | (e.g. any version that is not 1.8) |
| 21:56.45 | DrkShadow | 1.8?? I didn't even know there IS a v1.8.. |
| 21:56.56 | Qwell | keep up :p |
| 21:56.58 | nny | looks like linux itself told asterisk "f you, no more open files for you" |
| 21:57.11 | Qwell | DrkShadow: http://lists.digium.com/mailman/listinfo/asterisk-announce |
| 21:57.13 | Qwell | Go subscribe. |
| 21:57.41 | DrkShadow | no. I don't have time to administrate an Asterisk server full-time. |
| 21:57.56 | nny | paaart tiiiime lover! |
| 21:58.25 | Qwell | DrkShadow: 6-8 emails per month is too much for you? |
| 21:58.45 | DrkShadow | shrugs |
| 21:58.45 | DrkShadow | doing anything with them is. |
| 21:58.55 | Qwell | I would happily wade through asterisk-users every day, to make sure my system didn't have security issues. |
| 21:59.00 | DrkShadow | Asterisk 1.4.26.1 |
| 21:59.10 | nny | I subscribed, do I get a ham every month? |
| 21:59.25 | DrkShadow | I'd know within about 4 days if we had a security issue... |
| 21:59.31 | DrkShadow | 4 days at the latest. |
| 22:01.20 | Qwell | DrkShadow: The version you're running is vulnerable. |
| 22:01.25 | DrkShadow | to? |
| 22:01.40 | Qwell | a remote DoS |
| 22:01.48 | DrkShadow | whoopdy fuck. ;-) |
| 22:01.59 | DrkShadow | Last I checked there were no real security vulnerabilities. |
| 22:02.12 | Qwell | rolls his eyes |
| 22:02.25 | Qwell | That is a critical security vulnerability. |
| 22:02.36 | Qwell | AST-2009-010 |
| 22:02.36 | DrkShadow | that.. they.. can stop us from making/receiving phone calls.. |
| 22:02.46 | DrkShadow | remote crash.. |
| 22:02.53 | DrkShadow | looks |
| 22:03.26 | DrkShadow | pretty sure I've read that one.. |
| 22:03.26 | jdoe | lol |
| 22:04.19 | DrkShadow | On the other hand, 2010-002... |
| 22:04.38 | jdoe | not that I know anything about the specifics of that advisory, but remote corruption -> crash is often exploitable (whether someone has proven it or not), upgrading is ... a good idea. |
| 22:05.56 | nny | well that was a new one for me, time to figure out how many files linux allows a specific excetubale to open and increase it |
| 22:06.27 | jdoe | nny: ? |
| 22:06.32 | p3nguin | There isn't typically a limit unless a limit is set. |
| 22:06.47 | DrkShadow | nny: ulimit, I believe |
| 22:06.50 | nny | jdoe: http://lists.digium.com/pipermail/asterisk-users/2006-April/147204.html |
| 22:06.52 | p3nguin | man ulimit |
| 22:06.54 | nny | DrkShadow: yup |
| 22:07.00 | *** join/#asterisk saxa (~sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 22:07.10 | nny | must be an selinux thing |
| 22:07.26 | p3nguin | SElinux is a pain in everyone's ass. |
| 22:07.35 | nny | p3nguin: sharp pain in the ass |
| 22:08.04 | nny | yeah oddest part was consolesaid.. nothing. Popped open log, bam, chan_sip.c was bitching it couldn't even create the file. |
| 22:08.44 | nny | fixing now. happy friday everyone, hope you troubleshooting is less interesting |
| 22:08.49 | nny | you/your |
| 22:09.01 | nny | and or/saturday for those across the globe |
| 22:14.17 | *** join/#asterisk fofware (~Fabian@host215.186-110-114.telecom.net.ar) |
| 22:16.41 | *** part/#asterisk nny (~Scott@174.107.201.103) |
| 22:17.30 | *** join/#asterisk ariel_ (~chatzilla@c-98-254-138-7.hsd1.fl.comcast.net) |
| 22:21.00 | ANurmi | what should I put in the [globals] section of extensions.conf associated with OUTBOUNDTRUNK= for Dahdi system? |
| 22:21.28 | ritzt3ch | so Astersik engineer said this Remote end ringing WAS a back at a eariler release of astersik |
| 22:22.59 | ritzt3ch | but not this release that im at .... But I noticed on the CALLS that Fail is ONLY when im using more then 2 channels at once AND what fails is DTMF 8.9.0.1 DIALTONE (FAILS) what Works is DIALTONE DTMF 8.9.0.4 so it seems like it HAS to wait for dialtone to work |
| 22:25.35 | ANurmi | also the Asterisk CLI is throwing " app_dial.c:1750 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)" |
| 22:26.03 | ChannelZ | Anyone know of another util like sox but that can read multichannel AIFF filess? |
| 22:26.17 | pabelanger | ANurmi: *CLI> module load chan_dahdi.so |
| 22:28.10 | ANurmi | pabelanger: WARNING[23897]: loader.c:787 load_resource: Module 'chan_dahdi.so' already exists. |
| 22:28.52 | pabelanger | ANurmi: Then collect some debug, and pastebin the result. |
| 22:36.54 | *** join/#asterisk [canniballllera] (~cannibale@200-180-108-198.fnsce703.dsl.brasiltelecom.net.br) |
| 22:37.17 | *** part/#asterisk [canniballllera] (~cannibale@200-180-108-198.fnsce703.dsl.brasiltelecom.net.br) |
| 22:37.22 | saxa | hello, to register one box to another by iax2 is enough to put a register => line in the iax.conf file on one side, and on the other side is needed to create a context [box1] in iax.conf |
| 22:37.27 | saxa | correct ? |
| 22:37.29 | citywok | ruh roh, * 1.8 calendaring with exchange web services causes asterisk to crash. lol. make a test write and it tanks. |
| 22:37.58 | saxa | or do I need to have a [box2] conext also on the box1 iax.conf ? |
| 22:38.48 | saxa | and do I need to register with both boxes one to another if I want to exchange calls between them ? |
| 22:39.04 | a1fa | insource or outsource that is the question |
| 22:39.16 | *** join/#asterisk JParr (~JParr@24.244.133.116) |
| 22:41.40 | *** join/#asterisk Ziaeon (~Alchemist@adsl-074-166-171-132.sip.mia.bellsouth.net) |
| 22:43.00 | pabelanger | ~book |
| 22:43.00 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook, or http://ofps.oreilly.com/ |
| 22:43.04 | pabelanger | saxa: ^^ |
| 22:43.16 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 22:48.13 | JParr | anybody using a2billing? |
| 22:52.01 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 22:52.42 | *** join/#asterisk Juggie (~Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 22:55.45 | ritzt3ch | Think i found a Bug in asterisk but not sure how to pinpoint WHERE .... But I noticed on the CALLS that Fail is ONLY when im using more then 2 channels at once AND what fails is DTMF 8.9.0.1 DIALTONE (FAILS) what Works is DIALTONE DTMF 8.9.0.4 so it seems like it HAS to wait for dialtone to work |
| 22:56.46 | [TK]D-Fender | ritzt3ch: jsut make it wait longer |
| 22:57.37 | ritzt3ch | NOTICE[2897]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (3) Remote end Ringing |
| 23:00.09 | *** join/#asterisk fofware (~Fabian@host102.186-110-118.telecom.net.ar) |
| 23:00.45 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 23:00.46 | ritzt3ch | could .call files have anything to do with it |
| 23:01.16 | [TK]D-Fender | ritzt3ch: I'd do REAL testing to evaluate, including parallel tests with an analog phone |
| 23:01.28 | [TK]D-Fender | ritzt3ch: Change up the timing, etc |
| 23:04.14 | ritzt3ch | could it be waittime not set in my call file |
| 23:04.28 | ritzt3ch | i just have maxretries - 10 |
| 23:04.32 | ritzt3ch | retry time 30 |
| 23:04.36 | ritzt3ch | priority 1 |
| 23:04.43 | ritzt3ch | and chmod 777 |
| 23:05.36 | ritzt3ch | but its always fails when more then 2 ports dahdi/g0 is used and i hear dtmf 8.9.0.1 Then Dialtone on evey fail |
| 23:06.01 | ritzt3ch | shit because i just saw a Known bug that was fixed eariler on the version i had |
| 23:17.23 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 23:21.07 | beardy | noticed someone referenced his article/howto earlier and gets inspired to continue writing another :) |
| 23:22.36 | ManxPower | ritzt3ch, add a "w" or two at the beginning of the dialstring to make asterisk wait a few seconds. Remember Asterisk generally blind dials and cannot, does not wait for the dialtone |
| 23:26.41 | saxa | pabelanger: thx |
| 23:27.09 | ritzt3ch | thats what TKD kept on saying but ive tried that as that was the first thing he said |
| 23:27.19 | ritzt3ch | but IT CANNOT WAIT for dialtone is what your saying |
| 23:30.23 | saxa | pabelanger: http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/I_sect14_tt670.html <- thats ok when you have static ip on both sides |
| 23:30.46 | saxa | pabelanger: now, I have one box with the static IP and the other with the dynamic IP |
| 23:31.14 | saxa | pabelanger: how do I tell the static IP box where the dynamic IP one is ?? |
| 23:33.07 | pabelanger | saxa: you have the dynamic IP register to the static. |
| 23:33.22 | pabelanger | once registered, the static knows the dynamic IP |
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| 23:36.06 | ManxPower | ritzt3ch, you are correct Asterisk cannot wait for dialtone. Asterisk can easily wait a defined time before dialing. Dial(DAHDI/g1/wwww5551212) |
| 23:36.13 | ManxPower | that waits 2 seconds before dialing 5551212 |
| 23:36.24 | pabelanger | saxa: And if you look at the example in the book, they aren't setup for static IPs, becuase host=dynamic. |
| 23:36.42 | *** join/#asterisk marksaitis (~Mk@78-61-148-80.static.zebra.lt) |
| 23:36.45 | marksaitis | hey |
| 23:36.56 | ritzt3ch | as each digit is .5 seconds ive even tried dahdi/g0/wwwwwwwww and it still failed |
| 23:37.03 | marksaitis | does asterisk 1.8 have free text to speach feature? |
| 23:37.03 | marksaitis | ;] |
| 23:37.06 | ManxPower | then you have some other issue |
| 23:37.52 | ManxPower | What do you HEAR when you use the w's |
| 23:38.46 | *** join/#asterisk c0rnoTa (~c0rnoTa@nas-17.emserv.ru) |
| 23:39.50 | ManxPower | ritzt3ch, You should also try increasing your txgain. Maybe your transmit level is too low for the far end to detect. You can also try decreasing your txgain. Maybe you are transmitting so loud it is distorted. Also look at toneduration in chan_dahdi.conf. Maybe your DTMF tones are too short for the far end to detect |
| 23:40.23 | ManxPower | Asterisk's default tone duration is a little low for some systems to detect reliability |
| 23:40.52 | p3nguin | There were some typical recommended values for those two settings, but I don't remember them. |
| 23:41.03 | ritzt3ch | seems like a good idea hmmmm only happens unless i use more then 2 ports so i DIDnt thing of using the Txgain... |
| 23:41.09 | ritzt3ch | is that like interdigit timing |
| 23:41.36 | ManxPower | I recommend 0 for rxgain and txgain, or decrease by 2. I like 50 - 70ms tone duration. |
| 23:41.39 | ritzt3ch | when i had the butt set in For what i definely know is Calls that do fail sent DTMF before the system returned dialtone |
| 23:41.47 | pabelanger | marksaitis: app_festival |
| 23:42.02 | ManxPower | ritzt3ch, even after you wait for all that time? |
| 23:42.06 | ManxPower | with the w's? |
| 23:42.28 | ritzt3ch | i dont belive it worked but i didnt Hear it visually i should of actually tested that on the wwwwww |
| 23:42.50 | ritzt3ch | but they said it was still failing once i waited so long it was like 8 seconds |
| 23:42.58 | ManxPower | Also check the punchdowns. It is easy to accidentally connect 1 wire from one pair to 1 wire of a different pair. |
| 23:43.03 | ritzt3ch | but it for some reason didnt go off hook right |
| 23:43.05 | *** part/#asterisk c0rnoTa (~c0rnoTa@nas-17.emserv.ru) |
| 23:43.14 | ritzt3ch | it should be just rj11 to rj11 |
| 23:43.27 | ManxPower | your telco installed that rj11? |
| 23:43.32 | ritzt3ch | thats true like a bleeding pair |
| 23:43.46 | marksaitis | pabelanger, where do I get one? |
| 23:44.07 | ManxPower | We had something similar where 1 line outof the two worked. turns out one of the tails from one of the punchdowns was not removed and was touching the 2nd pair |
| 23:44.26 | ritzt3ch | actually my tech and i are Mitel dealers |
| 23:44.39 | ManxPower | cool, so you know about punchdowns 8-) |
| 23:45.22 | ritzt3ch | hmmm that is interesting thing you pring up though because 4 ports in use only if im doing an ALL-Page and if the common fail is because more then 2 3 could be failing due to a bad wire like you were mentioning |
| 23:45.54 | ritzt3ch | and it always seems to fail on when 3 or 4 are in use |
| 23:46.22 | ManxPower | ritzt3ch, make sure you have the power plugged into the analog card. I think modern cards refuse to work at all if you don't, but the older ones did not. |
| 23:46.25 | ritzt3ch | though i did think i asked him to swap wires around and it would still happen when i had him unplug the cable from dahdi/3-1 |
| 23:46.50 | ManxPower | ritzt3ch, what state are you in? |
| 23:47.17 | ritzt3ch | i did plug it in haha i had to get a special one because the new boxes are in Sata style |
| 23:47.22 | ritzt3ch | Im in NV-LSV |
| 23:47.30 | ritzt3ch | my tech is in So-CAl |
| 23:48.07 | marksaitis | Does anyone know where to get that app_festival? |
| 23:48.08 | marksaitis | ;] |
| 23:48.22 | ManxPower | In my area NY/NJ (well the area where we service, I'm not anywhere near there) Verizon proves that incompetence really does look like malice. |
| 23:48.37 | ritzt3ch | im thinking about giving up and maybe using Sip-Trunks but im having a issue on the outbound trunks |
| 23:48.49 | ritzt3ch | haha nice |
| 23:48.52 | ManxPower | ritzt3ch, analog sucks no matter what you do. |
| 23:49.12 | underdog | I'm messing around with some old cisco ATA 186's...have you ever seen an issue where they don't hang up the call when the remote end disconnects? |
| 23:49.13 | ritzt3ch | serious i built a 16 port FAX server that took alot of exhaustion |
| 23:49.54 | ritzt3ch | and we have customers that are mitel that like certain projects like advanced IVRs Faxing / Paging / Confercing / Email stuff |
| 23:51.34 | ritzt3ch | but maybe theres a bad port or the modules for port 3 and 4 are not in all the way .... but i wouldnt know why 2 of the ports wont work well |
| 23:51.47 | ManxPower | I don't mean to be mean, but you were not using Digium cards. |
| 23:53.17 | ritzt3ch | i have a tdm410p |
| 23:53.20 | ritzt3ch | 4 port |
| 23:56.41 | ritzt3ch | and the weirdest thing i see from time to time is -- Hungup 'DAHDI/3-1' |
| 23:56.41 | ritzt3ch | <PROTECTED> |
| 23:57.21 | ritzt3ch | i dont have a port 5-1 so im not sure if something is looped somewhere software wise |
| 23:57.44 | ritzt3ch | i only have 1-1 - 1-4 i could probally comment it out in /etc/dahdi/system.conf |
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