| 00:01.01 | [TK]D-Fender | hmmhesays, Nah.. I don't even use my E815 since I upgraded naturally. |
| 00:07.34 | hmmhesays | I got two months on my contract so I opted to wait and see if I could make this damn thing work |
| 00:07.41 | hmmhesays | so far so good |
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| 00:58.08 | Yourname` | Are you guys on Telus? |
| 00:58.23 | tengulre | Happy New Year ,everyone! |
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| 02:14.30 | dacs | is the is audio asterisk ebook for ipod |
| 02:14.31 | MrFollies | Does anyone here have any experience with Sipura devices? |
| 02:14.56 | dacs | MrFollies: for configuration |
| 02:15.00 | MrFollies | No |
| 02:15.28 | MrFollies | I am trying to make a direct IP call to my sipura (i.e. without asterisk) |
| 02:16.43 | dacs | MrFollies: i think you cofig your SIP info in the ATA , then you use like ipkall to asign you a number |
| 02:18.08 | dacs | me myself i wanted to do that , but decided to learn and understand asterisk |
| 02:19.10 | dacs | ~books |
| 02:19.20 | dacs | ~book |
| 02:19.21 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 02:19.41 | dacs | ~Tell MrFollies book |
| 02:19.43 | MrFollies | I am trying to figure out how to make my DID provider call the sipura directly rather than asterisk. I need to do this because I want to take asterisk offline for a day to upgrade it. |
| 02:21.09 | dacs | brb |
| 02:21.45 | MrFollies | To start with I have SJphone running on the same lan. I can call from sipura to SJphone but not the other way around. |
| 02:22.09 | MrFollies | I don't even see anything in the debug/syslog window. |
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| 02:42.03 | De_Mon | MrFollies you register the phone to the provider instead of asterisk. It's that simple. |
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| 02:49.53 | MrFollies | Provider is an inbound service only. It does not require registration. |
| 02:50.34 | MrFollies | A simpler question: why can't I make a call from SJ Phone to Sipura on the same lan. |
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| 02:54.21 | dacs | De_Mon: do you know if there is * audiobooks please |
| 02:54.35 | De_Mon | ive never tried to connect two phones togeather without some sort of authentication provider |
| 02:54.39 | De_Mon | dacs nope |
| 02:56.10 | dacs | De_Mon: nope there is or nope you don't know |
| 02:57.31 | russellb | i have never heard of one |
| 02:57.46 | MrFollies | Figured it out. I was not using sip: at the front of the dial string. |
| 02:58.26 | De_Mon | the question was if I knew so nope i know not |
| 02:59.04 | dacs | what if i am blind |
| 03:00.00 | dacs | oh well i was just hoping that there is an audio book for asterisk |
| 03:00.01 | JT | use a screen reader? |
| 03:00.20 | dacs | JT: what if i am driving |
| 03:00.22 | dacs | lol |
| 03:00.51 | De_Mon | SWEET tower defense for the ds |
| 03:00.53 | JT | then can you not make the screen reader make an audio file first? |
| 03:02.29 | dacs | JT: do you know if its possable to convert pdf to audio |
| 03:02.40 | dacs | man i just want this so bad |
| 03:02.43 | dacs | :) |
| 03:02.56 | De_Mon | dacs you could always bring a light instead of the mp3 player |
| 03:04.19 | JT | dacs: there is text in the pdf |
| 03:04.25 | JT | and there is also a html version |
| 03:04.49 | dacs | JT: ?? |
| 03:05.08 | JT | was self explanatory |
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| 03:08.08 | dacs | JT: well you please explain |
| 03:08.26 | JT | explain what? |
| 03:08.38 | dacs | < JT> dacs: there is text in the pdf |
| 03:08.45 | dacs | < JT> and there is also a html version |
| 03:08.51 | JT | the pdf file contains text |
| 03:08.58 | JT | how bloody hard is that to understand? |
| 03:09.04 | JT | html is also text |
| 03:09.41 | *** part/#asterisk kraptv (n=ryan@magic.skylab.org) |
| 03:10.05 | dacs | and your point is? |
| 03:10.24 | JT | text is readable with a screen reader, pretty obvious |
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| 03:53.06 | tzafrir_home | dacs, look for pdftotxt or pdf2txt or something similar |
| 03:53.33 | tzafrir_home | pdftotext |
| 03:54.49 | tzafrir_home | part of xpdf-utils on my system |
| 03:55.36 | tzafrir_home | oh, this is about a specific book |
| 04:03.57 | De_Mon | hmmmm to sleep or watch stardust |
| 04:04.00 | De_Mon | choices choices |
| 04:05.01 | [TK]D-Fender | De_Mon, got a chick to watch it with? |
| 04:05.41 | russellb | or a dude, whatever works for you |
| 04:06.30 | De_Mon | nope just me tonight |
| 04:07.23 | [TK]D-Fender | De_Mon, Watched it before? |
| 04:09.54 | *** join/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 04:10.29 | harryv | hey. any good tools for simulating multiple calls at once? to test my agi-app.. |
| 04:11.13 | De_Mon | no, just skipped around a bit and it looks pretty crappy... |
| 04:11.15 | russellb | asterisk is pretty good for that :) |
| 04:11.57 | De_Mon | I do believe I will fall asleep watching this |
| 04:13.11 | harryv | doh. of course. |
| 04:14.24 | russellb | use a bunch of call files to connect a Local channel to your AGI ... |
| 04:14.38 | russellb | have the extension for the Local channel Answer() and then do whatever you want to do to simulate the call |
| 04:14.50 | russellb | send digits, play some sounds, whatever |
| 04:17.38 | De_Mon | Oooo sweeny todd didn't do so good |
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| 04:51.16 | bmcghee`home | for music on hold what are the wav requirements |
| 04:51.39 | russellb | 8 kHz mono |
| 04:53.05 | WilliamK | russell do you have mysql experience by chance? |
| 04:54.40 | russellb | some ... |
| 04:54.49 | russellb | about to pass out, though ... |
| 04:55.29 | WilliamK | do you know how I can restore a db using mysqlimport? |
| 04:55.36 | WilliamK | I'm having alot of issues |
| 04:55.50 | bmcghee`home | russllb i converted it to that and its not playing still |
| 04:55.56 | WilliamK | when I think I'm getting somewhere it takes me to -> |
| 04:57.47 | russellb | you might want to try #mysql |
| 04:57.59 | WilliamK | k |
| 04:59.20 | WilliamK | nm, got the command this time |
| 04:59.27 | WilliamK | finally found a good page using google |
| 05:00.09 | FuriousGeorge | support from digium is per server, not per contract/institution/individual, right? |
| 05:01.54 | Corydon76-dig | Depends on what you're licensing. Generally, it's per card. |
| 05:02.05 | Corydon76-dig | for ABE, it per server |
| 05:02.40 | FuriousGeorge | Corydon76-dig: so if i use tdm400p its roughly 1000 USD per year per card i need help with? |
| 05:02.57 | russellb | no, for a tdm400p, it's free support |
| 05:03.06 | russellb | to a point, anyway |
| 05:03.07 | FuriousGeorge | im talking about software support |
| 05:03.34 | russellb | oh, then i have no idea how support is priced these days |
| 05:04.35 | Corydon76-dig | FuriousGeorge: No, ABE is considerably less nowadays |
| 05:04.40 | FuriousGeorge | russellb: perhaps there is a "silver package" where i just bribe you, and you persuade someone to take a look at my bug report |
| 05:04.43 | FuriousGeorge | ~ABE |
| 05:04.56 | FuriousGeorge | Asterisk Business Edition |
| 05:05.26 | russellb | jbot: ABE is Asterisk Business Edition, a commercial version of Asterisk from Digium, Inc. See http://www.digium.com/ for more information. |
| 05:05.27 | jbot | russellb: okay |
| 05:05.36 | Corydon76-dig | The original ABE package was $999, but the price points are lower, mostly because sales discovered that people wanted to buy a lower number of seats |
| 05:05.53 | russellb | Corydon76-dig: still is IIRC ... |
| 05:06.06 | russellb | Corydon76-dig: what you're talking about is the planned structure ... not yet implemented |
| 05:06.15 | Corydon76-dig | russellb: I believe only if you buy the full 120 seats |
| 05:06.20 | russellb | oh. |
| 05:06.38 | Corydon76-dig | They're now selling something like 30 seats at the base package |
| 05:06.54 | Corydon76-dig | I don't know the exact numbers, just generally |
| 05:07.36 | russellb | Corydon76-dig: i still don't think that's done ... according to store.digium.com, you can still only buy the full thing |
| 05:07.49 | russellb | so, shush you :-p |
| 05:09.58 | De_Mon | how does abe control the number of seats? |
| 05:10.20 | De_Mon | digium take the microsoft 'honor system' route? |
| 05:10.33 | Corydon76-dig | No, there's a license key |
| 05:11.32 | De_Mon | it locks down the number of sip/iax/zap entries or something? |
| 05:11.42 | Corydon76-dig | Number of concurrent calls |
| 05:11.48 | De_Mon | ooooo |
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| 05:14.12 | Corydon76-dig | Actually, I suspect if you wanted a 53-seat license of ABE, sales would figure it out and do it for you |
| 05:16.00 | russellb | I suspect that if a large sum of money ended up in my paypal account, I would happily look at a bug reported of the donor's choice. |
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| 05:16.18 | russellb | s/reported/report/ |
| 05:16.33 | FuriousGeorge | russellb: how big is big |
| 05:17.03 | FuriousGeorge | will WoW currency do? |
| 05:17.07 | russellb | lol |
| 05:17.16 | russellb | this is my job, i can't accept money for stuff like that ... |
| 05:17.39 | file | russellb: will you accept... a hug? |
| 05:17.43 | russellb | file: sure! |
| 05:17.48 | Corydon76-dig | But you could donate money to Freenode or another charity of Russell's choosing |
| 05:17.50 | file | I'm all out |
| 05:18.19 | [TK]D-Fender | say.... where has bkw_ been lately anyways? ;0 |
| 05:18.25 | Corydon76-dig | file: that's okay; I've got a fresh supply. I can lend a few to you... |
| 05:18.42 | Corydon76-dig | file: but you have to give some back sometime |
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| 05:19.13 | file | eep |
| 05:19.24 | russellb | FuriousGeorge: you have a bug that's buggin' you? |
| 05:20.08 | FuriousGeorge | http://bugs.digium.com/view.php?id=11557 has really got me down |
| 05:20.16 | FuriousGeorge | feelin blue |
| 05:21.34 | russellb | FuriousGeorge: if it is locking up on your, build with DEBUG_THREADS, and get the "core show locks" CLI command output when it locks up |
| 05:21.40 | russellb | and then someone can fix it |
| 05:21.50 | russellb | but first, put the digium card back in :-p |
| 05:22.25 | russellb | p |
| 05:23.22 | De_Mon | a used hug? grose |
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| 05:27.07 | MooingLemur | used hugs appreciate in value |
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| 06:02.41 | adelas | Hey, would anyone know if its possible to have voip go through a vpn tunnel? |
| 06:03.21 | Corydon76-dig | Yes |
| 06:03.21 | De_Mon | its... possible |
| 06:03.30 | adelas | well, i'm having a bit of a problem |
| 06:03.39 | Corydon76-dig | Not just possible; I've done it before |
| 06:03.43 | adelas | i have 2 sonicwalls with a site-to-site vpn connection |
| 06:03.57 | Corydon76-dig | Well, there's your first problem... sonicwall |
| 06:03.59 | adelas | the vpn tunnel is up, i can do a tftp to get things |
| 06:04.04 | adelas | and everything works fine |
| 06:04.08 | adelas | just the voip :( |
| 06:04.15 | Corydon76-dig | Use openvpn and everything will work perfectly |
| 06:04.20 | adelas | well this is all hardware |
| 06:04.26 | Corydon76-dig | Pffft |
| 06:04.38 | lucent | he's not joking, ipcop + zerina = bliss |
| 06:04.46 | Corydon76-dig | "Hardware firewall" is a marketing term |
| 06:04.47 | adelas | well the thing is i'm expanding network |
| 06:04.48 | lucent | (openvpn) |
| 06:04.51 | Corydon76-dig | There's no such thing |
| 06:04.53 | De_Mon | stardust ipcop ftw |
| 06:05.11 | adelas | well i don't have any computer running on this network beside the asterisk server |
| 06:05.42 | Corydon76-dig | Well, there is such a thing as a hardware firewall, but it doesn't contain electronics. A 4 ft thick concrete barrier is a hardware firewall |
| 06:06.01 | adelas | well, with a router |
| 06:06.05 | lucent | enough ridicule |
| 06:06.16 | adelas | any other recommendation |
| 06:06.18 | Corydon76-dig | There's no such thing as a hardware firewall when it comes to networking |
| 06:06.28 | adelas | router then |
| 06:06.38 | lucent | for SIP to work you need to pass UDP, TCP, etc. |
| 06:06.40 | Corydon76-dig | Use two Linux boxes with openvpn |
| 06:06.55 | adelas | i'm a noobie when it comes to linux |
| 06:07.10 | Corydon76-dig | It's not that difficult to learn |
| 06:07.21 | adelas | well i want to keep this as simple as possible with harddware |
| 06:07.24 | Corydon76-dig | plus it works worlds better than that sonicwall crap |
| 06:07.30 | adelas | yea problay |
| 06:07.30 | De_Mon | ipcop makes it pretty simple |
| 06:07.34 | adelas | but thats what 'm stuck with |
| 06:07.42 | adelas | unless theres other hardware out there? |
| 06:07.48 | lucent | ipcop was the one clear winner out of all the different toasters I tried |
| 06:08.12 | De_Mon | stardust wasn't as bad as i thought it would be |
| 06:08.12 | Corydon76-dig | ipcop works |
| 06:08.22 | lucent | I would use it over a PIX 501 because the endpoint client is less of a pain in the ass |
| 06:08.35 | lucent | that's true of any solution using OpenVPN though |
| 06:08.37 | De_Mon | its what pan's labrynth and road to terabithia should have been |
| 06:09.07 | adelas | well is there any hardware device vpn you guys would recommend? |
| 06:09.08 | De_Mon | an actual fantasy movie! And now, I go off to sleeps |
| 06:09.13 | lucent | adelas: no. |
| 06:09.32 | lucent | cisco PIX 501 is the industry standard for wankers who cannot afford a real VPN concentrator |
| 06:09.48 | Corydon76-dig | In actuality, ipcop is just as much a hardware solution as sonicwall |
| 06:10.05 | lucent | ipcop's installation and setup is awful IMO |
| 06:10.14 | lucent | what happens if there's two NICs of the same driver? |
| 06:10.18 | lucent | that's my complaint |
| 06:10.25 | Corydon76-dig | Yeah, but it's just as much software as sonicwall |
| 06:10.27 | lucent | not the interface or its function once properly configured |
| 06:10.30 | adelas | well i just want a stable and decent vpn connection between these 2 places |
| 06:10.47 | adelas | so i can expand the phones on other side and use voip |
| 06:10.48 | Corydon76-dig | smoothwall works, too |
| 06:11.01 | lucent | smoothwall is super nice, but it's not libre |
| 06:11.10 | Corydon76-dig | In fact, I think ipcop was based on smoothwall |
| 06:11.13 | adelas | so i guess no ever got sonic wall vpns to work here? |
| 06:11.35 | lucent | Corydon76-dig: you're full of crap tonight, boss. :/ |
| 06:11.44 | Corydon76-dig | adelas: in fact, sonicwall is as likely as not to be the culprit when people have trouble with voip here |
| 06:12.21 | Corydon76-dig | lucent: openwall? I can't remember the name of the package |
| 06:12.46 | Corydon76-dig | lucent: ipcop is a fork of something else, I know that |
| 06:13.24 | adelas | then what can it be? i have one sonic wall as the main router thats controlling the phone system and phones plug into it, that works perfectly fine. on the other end is another sonic wall, with phones - on this end it can contact back to the other end with anything except the voip |
| 06:13.28 | lucent | I stand down, you're very close |
| 06:13.32 | lucent | we're both wrong |
| 06:13.40 | Corydon76-dig | lucent: bingo, SmoothWall, I remembered correctly: http://en.wikipedia.org/wiki/IPCop |
| 06:13.46 | lucent | history says Shorewall |
| 06:14.18 | lucent | Smoothwall and IPCop have very little to do with eachother today though |
| 06:14.27 | lucent | at least superficially |
| 06:14.32 | adelas | well i know this is not a place to trouble shoot my router, i'm just asking if anyone seen this problem with site-to-site sonicwall vpns |
| 06:14.36 | lucent | the install on Smoothwall is so nice |
| 06:14.50 | lucent | adelas: you have sonicwall stuff right? call them. |
| 06:15.11 | adelas | hahaha |
| 06:15.13 | lucent | that's why one buys sonicwall, so that sonicwall support will help them |
| 06:15.39 | adelas | the last time i tried that, the license already expired and they wanted a credit card for charging |
| 06:16.14 | Corydon76-dig | Yeah, in this case, paying more isn't necessarily better |
| 06:16.20 | lucent | whose liable for that? come on now, we're suggesting valid and community sustainable alternatives |
| 06:17.05 | lucent | for client and site-to-site openvpn, IPCop + Zerina add-on is user-friendly, stable, and libre |
| 06:17.30 | Corydon76-dig | openvpn is also one of the most cross-platform VPNs out there |
| 06:17.58 | Corydon76-dig | You can run the server and the client on all 3 major platforms, and a few more, too |
| 06:18.20 | lucent | Ford, Buick, and Chevrolet |
| 06:18.34 | lucent | what are the few more though? |
| 06:18.45 | Corydon76-dig | Solaris, FreeBSD, NetBSD |
| 06:19.21 | Corydon76-dig | In addition to OS X, Windows, Linux |
| 06:19.49 | Corydon76-dig | Wouldn't surprise me if there was a BeOS server/client, too |
| 06:19.50 | lucent | Corydon76-dig: about IPCop though, what happens in setup if two NICs use the same driver? |
| 06:20.10 | Corydon76-dig | lucent: I don't use it personally, but it should be fine |
| 06:20.34 | lucent | and we're suggesting this to guy? |
| 06:20.37 | Corydon76-dig | I like my command line firewall scripts |
| 06:20.51 | Corydon76-dig | but I wouldn't suggest that to most people |
| 06:21.28 | Corydon76-dig | It does give a great amount of control that you can't abstract very well with a GUI |
| 06:21.36 | lucent | if WRT54G routers weren't so crappy I might suggest grabbing a couple of those |
| 06:21.48 | lucent | throughput is just wretched though on their MIPS processors |
| 06:22.40 | lucent | adelas: what could it kill to pay for sonicwall support? |
| 06:23.13 | lucent | it's that, or find your time to get a replacement, IMO |
| 06:27.02 | adelas | it cost alot |
| 06:27.06 | adelas | to renew the contract |
| 06:27.11 | adelas | thats where they make money from |
| 06:27.44 | adelas | if its some setting, i will find it. i just want like a yes/no/maybe/how answer :( |
| 06:27.59 | adelas | i jsut can't think of anyhting wrong with this right now that would not make it work |
| 06:28.14 | adelas | because everything else through the vpn tunnel works fine |
| 06:28.35 | adelas | is there anything special about the voip UDP? |
| 06:38.51 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
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| 06:53.30 | RedStalker_Mike | hi all) |
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| 08:23.48 | *** join/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 08:25.31 | Alexandre_fr | Hello ! I have a question about Agentcallback login |
| 08:26.04 | Alexandre_fr | I saw that is deprecated |
| 08:26.36 | badcfe | when i place a call in ami, is it possible to create it with a channel that is going to the asterisk itself? |
| 08:26.55 | badcfe | i want both call-legs to be handled in the dialplan |
| 08:27.25 | badcfe | is it good practice to just do a SIP/to_myself ? |
| 08:27.40 | badcfe | or will it lead to a loop? |
| 08:27.52 | Alexandre_fr | I tried to configure a login logout in the dialplan, but I don't know how to use the agent.conf |
| 08:27.53 | badcfe | i mean "loop detection"? |
| 08:28.10 | Alexandre_fr | so my question it's is possible ? :) |
| 08:29.46 | Alexandre_fr | or I have to use th Voice Mail password now ? |
| 08:33.55 | badcfe | anyone has an example of how to place a call to * itself and control what context it goes to? how should i configure the peer? |
| 08:46.42 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
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| 08:47.49 | ronr_ | hi, I'm having a lot of trouble getting my E1 to talk to asterisk, the telco now told me CSC should be point to point, not point to multipoint, where do I set this? |
| 08:48.18 | *** join/#asterisk stimpie (n=stimpie@84-104-5-115.cable.quicknet.nl) |
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| 09:32.46 | badcfe | how do i prevent getting loop detected when i call SIP/myself ? |
| 09:33.16 | mosty | why are you calling yourself in the first place? |
| 09:33.18 | badcfe | i really need to place a call to myself cause i use ami and need dialplan treatment ... |
| 09:33.50 | mosty | can you elaborate on that? |
| 09:34.10 | badcfe | i do AMI originate action, and i need both call-legs to be handled by dialplan contexts |
| 09:34.54 | mosty | but why do you need one sip client to call itself? |
| 09:35.32 | badcfe | because * is supposed to call a user and then dial another user for him, bridging them |
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| 09:36.40 | *** part/#asterisk bintut (n=bintut@203.125.63.150) |
| 09:37.27 | mosty | badcfe, you don't need to call yourself to do that. pastebin your originate command, i think you're doing something silly |
| 09:37.28 | badcfe | i need dialplan logic to control whats going on on both call-legs, while in an AMI originate there is only one context specified, while for the other call-leg one specifies a channel (on this channel, i need to control with dialplan logic in order to route it rigth etc) |
| 09:37.50 | mosty | use chan_local |
| 09:38.03 | badcfe | mosty: my current originate action is mainstream. |
| 09:38.18 | badcfe | mosty: i think chan_local is what im looking for |
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| 09:39.38 | badcfe | my current originate has "Channel: SIP/thingy@dingy", could i change it into "Channel: local/thingy" |
| 09:39.43 | badcfe | ? |
| 09:40.40 | mosty | badcfe, in most cases channel would be your sip account, and context would be some outgoing context which does not call your sip account |
| 09:41.41 | badcfe | mosty: thanks. from here i think ill be able to step further. |
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| 09:53.04 | Alexandre_fr | Somebody knows about using the agent.con inside the dialplan without using agentgallbacklogin (because it's deprecated) |
| 09:54.02 | Alexandre_fr | or I have to use the VoiceMail authentification ? |
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| 09:58.16 | mosty | Alexandre_fr, what are you tying to do? |
| 09:58.19 | mosty | trying, rather |
| 09:59.35 | Alexandre_fr | I want agent, with a login logoff |
| 10:01.45 | mosty | it appears that the best thing to do is use realtime queue members |
| 10:02.37 | Alexandre_fr | it's with addqueuemember and removequeuemember ? |
| 10:04.32 | mosty | it's worth a try |
| 10:06.27 | Alexandre_fr | I already did something with that, but I want the agents to give a password, so i think I have to use VMauthenticate for that ? |
| 10:07.18 | mosty | just use Background or Read |
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| 10:07.25 | *** part/#asterisk bintut (n=bintut@203.125.63.150) |
| 10:12.09 | Alexandre_fr | And if I want use "autologoff= " which is in agent.conf, I have to do it in the dialplan ? |
| 10:20.32 | *** join/#asterisk codn (n=codaine@88.240.61.136) |
| 10:21.16 | *** part/#asterisk mitcheloc (n=mitchel@adsl-68-120-227-239.dsl.irvnca.pacbell.net) |
| 10:21.31 | badcfe | when dialing some Local/thingy, may i pass dialplan variables? |
| 10:21.32 | codn | hi, can anyone help me with a nat issue? |
| 10:22.05 | badcfe | i mean .. i want my channel variables to be inheritent into the target local-channel. |
| 10:23.03 | mosty | badcfe, pastebin your originate command |
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| 10:42.40 | MagInfo | Hi there |
| 10:47.11 | Porks | hi |
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| 10:53.17 | codn | is there a way to disable native bridging between specific clients? |
| 10:54.03 | MagInfo | codn: Just set nat = no and canreinvite = yes |
| 10:54.31 | codn | but one client is behind a nat, is that ok? |
| 10:54.40 | tzafrir_laptop | generally: force Asterisk to process the data stream |
| 10:55.15 | tzafrir_laptop | For instance, if there's some transcoding done by Asterisk, there can't be native bridging |
| 10:55.45 | codn | but both devices are cisco ata 186s. so, they do support same codecs |
| 10:56.10 | tzafrir_laptop | I figure both support both ulaw and alaw |
| 10:56.20 | tzafrir_laptop | (g711u and g711a) |
| 10:56.23 | codn | probably |
| 10:56.47 | MagInfo | codn: And also both clients must have white addresses |
| 10:56.53 | tzafrir_laptop | But why do you want to disable it? |
| 10:57.04 | tzafrir_laptop | What I suggested means Asterisk does some extra work |
| 10:57.18 | codn | in fact the problem -i guess- is that asterisk assuming remote device on one of the localnets. (which is not in fact) |
| 10:57.21 | codn | ok heres the thing |
| 10:59.18 | *** join/#asterisk krdian_ (i=krdian@killer.radom.net) |
| 10:59.22 | codn | asterisk stays behind the corporate firewall. Its a DMZ host so can be reached from outside. (10.0.0.0/16 is the corporate wan). So, if I am to add another ata 186 to my home and try to call someone. What I got was having internal IP (say, 10.0.1.253) was passed to the ata @ my home |
| 10:59.50 | codn | so I can hear, but the remote side wont be able to |
| 11:00.23 | mosty | codn: you should be setting externip and localnet in sip.conf |
| 11:00.28 | codn | when I sniff the network, remote site sends RTP packets directly to me. But my ata 186 tries to send out to 10.0.1.253, which is not reachable from here obviously :P |
| 11:00.35 | codn | those are correct |
| 11:01.28 | codn | btw I'm using 1.2.13 provided with debian, is there a way that 1.2.13 processes localnets in a wrong way? or assumes 192.168.0.0/16 all local? |
| 11:01.40 | *** join/#asterisk c0re| (n=c0re@spray.anyhost.ru) |
| 11:01.48 | mosty | codn: did you try what i said? |
| 11:02.23 | c0re| | Someone can help to understand with CDR ? |
| 11:02.57 | codn | mosty: I have em correctly set up |
| 11:03.14 | mosty | codn, pastebin them, prove it |
| 11:03.16 | codn | an external 7960 phone can call anyone with no problems at all |
| 11:03.29 | codn | :) k |
| 11:03.59 | c0re| | 2007-12-25 18:40:50","2007-12-25 18:40:50","2007-12-25 18:57:45",1015,1015,"ANSWERED","DOCUMENTATION" |
| 11:04.38 | c0re| | start time 18:40:50 and end time ring 18:57:45 And the person has answered ? |
| 11:04.51 | codn | http://codaine.pastebin.ca/833549 |
| 11:05.43 | c0re| | Why two times this record? ( 2007-12-25 18:40:50 ) ?? |
| 11:06.12 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:06.38 | mosty | codn, that pastebin site uses broken javascript or something, it looks blank with firefox :( |
| 11:07.03 | *** join/#asterisk waKKu (n=eurulo@unaffiliated/wakku) |
| 11:08.37 | codn | mosty: got my private msgs? |
| 11:09.13 | c0re| | Someone can help me? |
| 11:09.54 | mosty | codn: yeah. next step would be to run tshark to see what asterisk is sending out. if possible run it on your router at work and at home |
| 11:10.37 | codn | I can get SIP messages on asterisk's console |
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| 11:19.14 | codn | mosty: http://pastebin.com/d5d3a1a4 |
| 11:19.20 | af_ | for music on hold, a sound adapter is need on the pc? |
| 11:19.47 | codn | this is yesterdays log. I've marked what I see imprtant with a bunch of ****s |
| 11:19.51 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 11:21.23 | codn | I mean o=... 10.0.0.253 and c=... 10.0.0.253 lines, should have been the servers address, since they match localnet |
| 11:21.47 | mosty | codn: what version of asterisk? |
| 11:22.07 | codn | Asterisk 1.2.13 built by pbuilder @ grnetbox on a x86_64 running Linux on 2007-11-29 22:50:52 UTC |
| 11:22.30 | mosty | codn: is that a debian etch box? |
| 11:22.31 | codn | ts a debian etchy |
| 11:22.35 | codn | yea |
| 11:22.36 | mosty | ah |
| 11:22.50 | codn | should I switch to 1.4? |
| 11:23.06 | mosty | depends, how new is your install? |
| 11:23.36 | codn | its been running for 6 mnths. but I can reconfigure extensions and all others if needed |
| 11:23.43 | codn | will 1.4 upgrade fix this? |
| 11:25.08 | mosty | codn: i'm not sure. if it was a brand new install i'd say try 1.4 and see how it goes |
| 11:25.24 | mosty | externip/externhost + localnet should sort that stuff out |
| 11:27.00 | *** join/#asterisk myiagy (n=Jose@200.215.59.133) |
| 11:27.15 | codn | i c, maybe thats ata 186s misconfiguration |
| 11:30.29 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 11:32.59 | tzafrir_laptop | codn, you have backported 1.4 debs at http://pkg-voip.buildserver.net/debian |
| 11:33.16 | codn | ah thnx I was looking 4 em |
| 11:35.47 | ronr_ | hi, I'm having a lot of trouble getting my E1 to talk to asterisk, the telco now told me CSC should be point to point, not point to multipoint, where do I set this? |
| 11:36.03 | codn | can I just use the same sip.conf with 1.4? |
| 11:38.43 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
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| 11:47.25 | *** part/#asterisk c0re| (n=c0re@spray.anyhost.ru) |
| 11:50.47 | Alexandre_fr | somebody understand AEL ? |
| 11:50.50 | Alexandre_fr | 6121 => { 301 &queue-addremove(dispatch,10); 302 &queue-success(); 303 } |
| 11:50.52 | Alexandre_fr | oops |
| 11:51.09 | Alexandre_fr | 6121 => { |
| 11:51.30 | Alexandre_fr | &queue-addremove(dispatch,10); |
| 11:51.34 | Alexandre_fr | &queue-success(); |
| 11:51.38 | Alexandre_fr | } |
| 11:52.21 | Alexandre_fr | so I understood that: when you call 6121 it's go to the macro queue-addremove |
| 11:53.30 | Alexandre_fr | but it goes to macro queue-success anyway , or only if the first macro finish well ? |
| 11:54.02 | Alexandre_fr | it's from here |
| 11:54.03 | Alexandre_fr | http://svn.digium.com/view/asterisk/team/oej/minivoicemail-1.4/doc/queues-with-callback-members.txt?revision=89663&view=markup |
| 11:54.06 | Alexandre_fr | line 300 |
| 11:55.51 | Alexandre_fr | i come back in 1 hour |
| 12:02.07 | mosty | codn: i believe so, for the most part |
| 12:08.13 | codn | thers heavy traffic atm. so I'll upgrade tonight. Btw, that thing to disable native bridging did not work |
| 12:08.24 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
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| 12:32.02 | codn | tzafrir_laptop: dependencies of the backport doesnt fit etch's packages |
| 12:32.38 | tzafrir_laptop | codn, could you please give an example? |
| 12:32.48 | codn | sure holdon |
| 12:32.54 | mosty | codn, is it zaptel? |
| 12:33.04 | codn | i guess one was libspeex |
| 12:34.08 | codn | k, it depends on libspeex2 >= 1.2~beta2-1 and libtone1 >= 1:1.4.5 |
| 12:34.22 | codn | not on etch's or backports' repositories |
| 12:35.52 | mosty | codn: apt-get install libtonezone1=1:1.4.5, and a similar command for libspeex2, should fix that. find the exact versions in apt-cache show libtonezone1 and apt-cache show libspeex2 |
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| 12:36.12 | shadebob | hi, |
| 12:37.09 | codn | mosty: theres no libtonezone1=1:1.4.5 in debian or buildserver.net repos |
| 12:38.27 | mosty | codn: it's in tzafrir's repository. find the exact version with apt-cache show libtonezone1 |
| 12:39.47 | codn | none. http://pkg-voip.buildserver.net/debian/pool/main/, thers not even a libt directory :b |
| 12:40.02 | codn | and no libspeex2 in http://pkg-voip.buildserver.net/debian/pool/main/libs/ |
| 12:40.12 | codn | neither debian repos have 'em |
| 12:40.18 | mosty | libtonezone1 should be in the zaptel dir |
| 12:40.45 | mosty | try http://updates.xorcom.com/rapid/ |
| 12:43.16 | codn | found the file, I think you should rebuild Packages* files at http://pkg-voip.buildserver.net/debian/dists/etch/main/binary-i386 |
| 12:43.24 | codn | no definition there |
| 12:46.42 | *** join/#asterisk marzubus (n=keganh@84-74-184-161.dclient.hispeed.ch) |
| 12:57.13 | __freedom__lover | can someone tell me if it's mandatory use 2 ips in a stun server? |
| 13:05.16 | *** join/#asterisk benjamoon (n=benjamoo@aqu33-2-82-225-97-35.fbx.proxad.net) |
| 13:08.27 | anonymouz666 | __freedom__lover it is. |
| 13:08.39 | __freedom__lover | cpm: did i ask a stupid question? |
| 13:08.39 | benjamoon | Hi there, I've just installed Asterisk and am going through the doc but I'm having trouble finding anything about what I want to do... I need to call my SIP number (set up on Asterisk as the outbound service) from an external line and somehow attach or define another external number that I want Asterisk to call/forward via the SIP service... the idea being to call international numbers cheaply through my SIP rather than full-pr |
| 13:08.39 | benjamoon | ice from my mobile :) Does this make sense and is it possible? |
| 13:10.55 | Alexandre_fr | yes it's possible |
| 13:11.57 | codn | benjamoon: might u b looking for http://www.ipkall.com/ ? |
| 13:12.04 | codn | or a DID service? |
| 13:12.54 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:18.04 | benjamoon | codn: not really, I'm looking to forward calls via my current SIP through an asterisk set up on a home server |
| 13:18.22 | benjamoon | Alexandre_fr: can you point me towards the right doc? |
| 13:18.30 | mosty | benjamoon, do you want to make calls to the regular telephone network? |
| 13:18.50 | benjamoon | yep |
| 13:19.17 | mosty | benjamoon, then you will need a telephone line, or a voip termination service that does interface with the regular telephone network |
| 13:19.20 | benjamoon | from a regular mobile -> asterisk logged on to my SIP -> the other external phone |
| 13:19.40 | shido6 | do you have a sip account with someone? |
| 13:19.42 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:19.43 | benjamoon | yes |
| 13:19.57 | mosty | benjamoon, have you read the book? |
| 13:19.58 | mosty | ~thebook |
| 13:19.59 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com |
| 13:20.57 | benjamoon | I'm going through the doc now but I'm having trouble formulating what I want to do, searching for 'maybe forwarding external calls through SIP' isn't coming up with much :-) |
| 13:22.26 | shido6 | hehe |
| 13:22.49 | shido6 | you want to make a call from your regular mobile to your sip account to an external phone on the PSTN or on a voip network ? |
| 13:23.09 | shido6 | tell us what you want to do leaving any Asterisk specific words out |
| 13:23.31 | *** join/#asterisk RockHound (n=rockhoun@85.183.138.242) |
| 13:23.31 | codn | if I got it right, he wants to receive pstn calls and forward to pstn again |
| 13:23.38 | mosty | benjamoon, you need to setup the sip account in asterisk (easy), then get some hardware to connect your pc to your telephone line (digium/sangoma card, or an ata maybe- easy), then setup your dialplan for send calls from the telephone line to some number via the sip server |
| 13:23.39 | codn | to cut costs |
| 13:23.39 | RockHound | good day |
| 13:24.05 | RockHound | anyone with mISDN knowledge in here currently? |
| 13:24.57 | mosty | RockHound, i've used it a few times but my advice is to avoid hardware that only works with misdn |
| 13:25.03 | codn | or w/o any hardware, get a DID account to receive pstn calls from ordinary phones. And then get a SIP-out account to call pstn numbers |
| 13:26.14 | RockHound | mosty: works great except that I can not get my outgoing call to send my callerID even if I force it |
| 13:26.18 | mosty | it's probably cheaper to just buy a calling card |
| 13:26.24 | benjamoon | shido6, yep that's about right - bascially I want Asterisk to receive an incoming call which I make to my SIP number (that bit is working fine and my softphone rings) but if I add a certain extention then Asterisk calls an external regular number through my SIP account and connects me to that external number |
| 13:26.43 | mosty | RockHound, that's something you will need to check that your telco supports |
| 13:27.14 | shido6 | use pastebin.ca and show us your dialplan |
| 13:28.06 | RockHound | mosty: the telco supports it as the pbx which is being replaced worked fine |
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| 13:40.56 | *** part/#asterisk RockHound (n=rockhoun@85.183.138.242) |
| 13:42.16 | zver | hello. i update zaptel and asterisk version. when i start asterisk: ERROR[24092]: chan_zap.c:7218 mkintf: Unable to get span status: Inappropriate ioctl for device. Why? How i can fix it? |
| 13:43.03 | zver | before update all worked fine |
| 13:59.45 | *** join/#asterisk _ys (i=yuri@91.151.196.254) |
| 14:02.27 | *** join/#asterisk [gnubie] (n=[gnubie]@cm71.gamma182.maxonline.com.sg) |
| 14:02.31 | [gnubie] | hello all.. |
| 14:03.21 | [gnubie] | in using variables, can i also use ${EXTEN} to call for voicemail? |
| 14:04.39 | [gnubie] | for example, if i have a line like this: exten => _10X,n,VoiceMail(${EXTEN},u) |
| 14:04.49 | mosty | gnubie, depends what you mean. ${EXTEN} will always be the extension you're at in whatever context you're in, in the dialplan |
| 14:05.15 | mosty | in that case, ${EXTEN} would be something starting with 10 |
| 14:05.34 | *** join/#asterisk l2trace99 (n=asd@75.112.133.235) |
| 14:05.55 | hmmhesays | and is 3 digits |
| 14:06.08 | [gnubie] | mosty: does it mean that if the extension called is 100, the ${EXTEN} will also be 100? |
| 14:06.45 | hmmhesays | ${EXTEN} variable is the currently extension that was dialed |
| 14:06.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 14:06.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:07.08 | mosty | what hmmhesays said. but if you Goto some other extension, ${EXTEN} will be that instead |
| 14:08.13 | [gnubie] | oh.. so that it, it will always match whatever extension is dialed |
| 14:08.44 | codn | tzafrir_laptop: I got ur packages installed and configured. but it seems now it doesnt have sip commands on CLI. am i missing something? |
| 14:09.08 | [gnubie] | because, it will be possible that even 100 is called, i can explicitly call the voicemailbox 101 instead |
| 14:09.38 | *** join/#asterisk RoyK (n=roy@ip-3-1-149-91.dialup.ice.no) |
| 14:09.57 | hmmhesays | yes |
| 14:11.04 | *** join/#asterisk a-s (i=root@84.247.50.111) |
| 14:12.16 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-162-97-18.bstnma.east.verizon.net) |
| 14:13.12 | a-s | I call dial from my dialplan, and when I call dial again from the same place to the same number, the first call dies («Auto fallthrough, channel 'SIP/89.38.172.141-08214b40' status is 'CONGESTION'»), and the second call dials. What shell I do to obtain busy to the second, and to keep alive the first call ? |
| 14:13.38 | tzafrir_laptop | codn, what happens if you run: module load chan_sip.so |
| 14:14.39 | codn | damn, unable to load sip.conf. lemme check. sorry :) |
| 14:14.53 | mosty | a-s: generally that's done on the sip server side. however you can use the GROUP_COUNT function. look it up on the wiki |
| 14:15.12 | tzafrir_laptop | zver, what versions of zaptel and asterisk? What is the output of: cat /sys/module/zaptel/version |
| 14:16.10 | zver | tzafrir_laptop: i find solve this bug - http://forums.digium.com/viewtopic.php?p=58787&sid=a6526c8a457f0071a7cddabd7f53512f |
| 14:16.36 | zver | i have 1.4 version. this bug exist before 1.4.4 |
| 14:16.48 | *** join/#asterisk ManxPower (n=manxpowe@28.sub-75-200-17.myvzw.com) |
| 14:17.20 | a-s | mosty: Thank you. I will test now. |
| 14:18.01 | *** join/#asterisk Michofreiha (n=michofr@mail.splendor.net) |
| 14:19.49 | [gnubie] | i want to compress my 10[1-9] extensions by using patterns but my 101 is a zap channel and i don't know if you guys will recommend this conversion => http://www.privatepaste.com/fa9ljGgwUj |
| 14:20.37 | [gnubie] | currently, i have 5 lines for each extension.. |
| 14:20.51 | tzafrir_laptop | zver, was your issue actually resolved? If not: I suspect either an older Zaptel is still loaded, or your Asterisk has been built with an older zaptel.h |
| 14:21.14 | ManxPower | [gnubie]: now you understand the importance of good dialplan planning. |
| 14:21.36 | [gnubie] | ;) |
| 14:21.42 | Michofreiha | hello All |
| 14:21.58 | ManxPower | there is no way to avoid dialing delays with your current dialplan |
| 14:22.08 | Michofreiha | Is there any channel where we can find answers regarding voip general questions? |
| 14:23.03 | [gnubie] | ManxPower: what do you suggest then? is the delay noticeable? |
| 14:23.06 | tzafrir_laptop | [gnubie], compress just 10[2-9]? Or be optimistic and rely on the fact that explicit numbers have priority over matched patterns? |
| 14:23.22 | hi365 | hi |
| 14:23.32 | hi365 | hello tzafrir_laptop |
| 14:23.36 | ManxPower | I would suggest not have overlapping patterns of differning length. |
| 14:23.38 | tzafrir_laptop | (and that you'll actually remember that next time you read your dialplan) |
| 14:23.41 | *** join/#asterisk ddunavant (n=David@66.170.97.28) |
| 14:23.43 | tzafrir_laptop | hi |
| 14:23.50 | [gnubie] | tzafrir_laptop: nice idea. i'll just compress the 2-9 |
| 14:23.55 | ddunavant | hello |
| 14:24.05 | hi365 | tzafrir_laptop: would it be ok to pm you with a support question? |
| 14:24.30 | *** part/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 14:25.17 | [gnubie] | thanks guys for your suggestions.. :) |
| 14:25.40 | mosty | whenever debconf asks me a question on amd64 installs, the terminal is all messed up, how can i fix this? |
| 14:26.05 | tzafrir_laptop | alt-f2 , alt-f1 ? |
| 14:26.09 | tzafrir_laptop | Is that Sarge? |
| 14:26.25 | hi365 | Im having a problem with a pri (Bezeq) that randomly resets. im using sangoma hardware, and ive double checked every posible option that google would sugest |
| 14:27.25 | ManxPower | hi365: define "randomly resets" |
| 14:27.37 | tzafrir_laptop | The resets are initiated by your side or by their side? |
| 14:27.42 | codn | mosty: ru still there? I upgraded to 1.4 and still have the same problem. |
| 14:28.07 | hi365 | I havent found any way to reproduce the problem (other than to wait - it usually happens at least once every 24-48 hours) |
| 14:28.12 | a-s | mosty: how can I use GROUP_COUNT for a SIP channel? |
| 14:28.14 | ManxPower | randomly resets could be "Reset channel x" message on the console, dropped calls, red alarm, yellow alarm |
| 14:28.17 | a-s | I do not understand... |
| 14:29.10 | hi365 | ManxPower: im not sure about alarms. resets means: drops all calls, and restarts asterisk |
| 14:29.44 | ManxPower | hi365: exactly what is the symptom of the reset? |
| 14:30.05 | ManxPower | Oh, so Asterisk basically randomly stops and then starts? |
| 14:30.36 | hi365 | ManxPower: yup. and doesnt leave any cluses in the log. |
| 14:30.59 | ManxPower | hi365: sounds to me like asterisk is crashing and being restarted by the safe_asterisk script. |
| 14:31.14 | *** join/#asterisk Atamido (n=atamido@cpe-70-116-1-195.austin.res.rr.com) |
| 14:32.25 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-121-28.pskn.east.verizon.net) |
| 14:32.34 | *** join/#asterisk mbranca (i=daemon@mi-gw1.voismart.net) |
| 14:32.34 | hi365 | ManxPower: hmm, thats sounds like a posibility. how do i go about troubleshooting? |
| 14:32.54 | ManxPower | hi365: what version of Asterisk are you using? |
| 14:33.11 | hi365 | 1.4 |
| 14:33.34 | ManxPower | That is really, really, really old. You should be using 1.4.16.2 |
| 14:33.43 | hi365 | ur i meant 1.4.11 |
| 14:33.57 | hi365 | and 1.4.16.2 at another site |
| 14:34.23 | ManxPower | Then your first thing to do is upgrade Asterisk. There are several denial of service attach vulns, including remote crashes in older Asterisks |
| 14:34.33 | russellb | there have been 559 changes made to asterisk 1.4 since 1.4.11 :) |
| 14:35.35 | hi365 | :) but as i said, im having the same problems at another site (1.4.16.2) |
| 14:36.10 | *** join/#asterisk af_ (n=getsmart@88-149-240-60.dynamic.ngi.it) |
| 14:41.20 | *** join/#asterisk brpvieira (n=bernardo@c9118288.static.bhz.virtua.com.br) |
| 14:41.33 | ManxPower | hi365: You should make sure Asterisk generates a core dump and that the OS allows Asterisk to generate a coredump. |
| 14:41.43 | tzafrir_laptop | hi365, did you check the voip-info page regarding zapata.conf? they mention some magic value regarding Bezeq |
| 14:42.58 | hi365 | tzafrir_laptop: yup. and it didnt seem to help |
| 14:44.13 | hi365 | resetinterval=never |
| 14:44.34 | *** join/#asterisk pepo-- (n=pepOSX@190.79.246.105) |
| 14:44.41 | hi365 | ManxPower: where can i find instructions? |
| 14:46.00 | ManxPower | hi365: your local Linux admin manual should be able to tell you. check the ulimit command. |
| 14:47.08 | hi365 | thanks |
| 14:50.28 | *** join/#asterisk BeeBuu (n=chatzill@219.135.40.228) |
| 14:50.50 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 14:51.18 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 14:53.41 | mocker | Anyone have a rough cost estimate of a PRI in the midwest? |
| 14:53.49 | *** join/#asterisk ChkDigit (n=mrw@static24-89-65-166.regina.accesscomm.ca) |
| 14:54.11 | hi365 | tzafrir_laptop: any other bezeq tips? |
| 14:54.22 | hi365 | how often id normal for the pri to restart? |
| 14:55.04 | hi365 | s/id/is/ |
| 14:56.37 | *** join/#asterisk Maliuta (n=nikolai@ppp214-92.static.internode.on.net) |
| 14:57.26 | dacs | good morning all |
| 14:57.54 | dacs | bbl |
| 14:57.59 | *** part/#asterisk dacs (n=haiger@unaffiliated/dacs) |
| 14:59.29 | Alexandre_fr | hey ! |
| 14:59.38 | a-s | if I obtain congestion, how can I wait a period of time, in which to set a periodic message, then to try to call again? |
| 14:59.48 | Alexandre_fr | i have a question about the queues |
| 15:00.39 | Alexandre_fr | exten => 1,1,Read(CLIENT_NUMBER,beep) |
| 15:00.39 | Alexandre_fr | exten => 1,n,Queue(support|t|www.${CLIENT_NUMBER}.com) |
| 15:00.39 | Alexandre_fr | exten => 1,1,Read(CLIENT_NUMBER,beep) |
| 15:00.39 | Alexandre_fr | exten => 1,n,Queue(support|t|www.${CLIENT_NUMBER}.com) |
| 15:00.39 | Alexandre_fr | exten => 1,1,Read(CLIENT_NUMBER,beep) |
| 15:00.39 | Alexandre_fr | exten => 1,n,Queue(support|t|www.${CLIENT_NUMBER}.com) |
| 15:00.41 | Alexandre_fr | exten => 1,1,Read(CLIENT_NUMBER,beep) |
| 15:00.43 | Alexandre_fr | exten => 1,n,Queue(support|t|www.${CLIENT_NUMBER}.com) |
| 15:00.45 | Alexandre_fr | exten => 1,1,Read(CLIENT_NUMBER,beep) |
| 15:00.47 | Alexandre_fr | exten => 1,n,Queue(support|t|www.${CLIENT_NUMBER}.com) |
| 15:00.48 | *** mode/#asterisk [+b %Alexandre_fr!*@*] by russellb |
| 15:00.51 | BeeBuu | hi,all. |
| 15:01.08 | tzafrir_laptop | jbot, tell Alexandre_fr about pb |
| 15:01.08 | hi365 | thanks russellb |
| 15:01.24 | russellb | Alexandre_fr: please use pastebin.ca ... |
| 15:01.29 | BeeBuu | is there any USB asterisk card now? |
| 15:01.33 | *** mode/#asterisk [-b %Alexandre_fr!*@*] by russellb |
| 15:01.36 | Alexandre_fr | sorry |
| 15:01.40 | BeeBuu | anyone know that? |
| 15:01.47 | Alexandre_fr | I didn't know my past was active |
| 15:02.52 | brpvieira | BeeBuu: not sure about USB cards, but there's astribank... |
| 15:02.54 | tzafrir_laptop | hi365, again, the issue is not that the PRI resets. The issue is that Asterisk crashed |
| 15:03.09 | BeeBuu | astribank? |
| 15:03.13 | hi365 | oh. now i get it |
| 15:03.20 | russellb | ~astribank |
| 15:03.21 | jbot | hmm... astribank is a USB-connected channel bank made by Xorcom with zaptel drivers. http://xorcom.com/astribank/features.html . Units of 8 FXS, 16 FXS and 8FXS/8FXO are availble. Drivers: http://xorcom.com/drivers/astribank/ . Ask tzafrir |
| 15:03.23 | Alexandre_fr | I would like to know how the agent can have the www.${CLIENT_NUMBER}.com |
| 15:03.53 | BeeBuu | O |
| 15:04.01 | BeeBuu | thanks guys. |
| 15:04.09 | brpvieira | BeeBuu: see... faster than I could search my favorites |
| 15:04.18 | BeeBuu | let me take a look... |
| 15:04.33 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 15:04.45 | brpvieira | you might want to have a look at some TDMoE interfaces as well |
| 15:05.19 | Alexandre_fr | thank you for pastebin.ca I didn't know this website |
| 15:07.03 | BeeBuu | ~TDMoE |
| 15:07.04 | jbot | rumour has it, tdmoe is for a situation where you need TDM reliability without traditional TDM hardware. TDMoE is useful because it allows the above familiarity, flexibility, and reliability of TDM but over inexpensive Ethernet instead of T1s or E1s. |
| 15:09.12 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 15:09.19 | *** join/#asterisk davidnicol (n=chatzill@198.247.174.254) |
| 15:14.34 | Havokmon | I have an asterisk connected to a Fujitsu 9600 via a PRI. From the asterisk I can dial 3xxx and get a Fujitsu extension. I have another PRI from the CO on the Asterisk. How could I redirect a block of DIDs (xxx-5xxx)on the asterisk to the Fujitsu (5xxx) easily? |
| 15:14.40 | Havokmon | Could I just do this: exten => 5XXX,n,Macro(dialout-trunk,2,${EXTEN},,) ? |
| 15:15.39 | ManxPower | Havokmon: I guess that depends on exactly WHAT the macro "dialout-trunk" is. We don't know. |
| 15:16.58 | hmmhesays | oh ms exchange is driving me insane today |
| 15:18.29 | ronr_ | hi, my telco tells me I need point to point, not point to multipoint, is that the signalling pri_cpe setting? |
| 15:18.30 | *** join/#asterisk CapRicORN^80 (n=you@207.176.6.56) |
| 15:19.12 | Havokmon | Hrm ok.. that must be a trixbox macro then. |
| 15:19.15 | CapRicORN^80 | hi i am installing asterisk-addons . configure and make was succesfull but getting error in make install |
| 15:19.38 | CapRicORN^80 | this is the error /bin/sh: -c: line 3: syntax error: unexpected end of file |
| 15:22.16 | hmmhesays | no matter what exchange gives me the same IVR |
| 15:22.19 | davidnicol | what does "sent into invalid extension 's' in context 'default', but no invalid handler" mean? I have created a s exten in the default context |
| 15:22.23 | CapRicORN^80 | http://pastebin.com/m62a36a33 |
| 15:22.37 | davidnicol | I am trying to dial out from new * by copying a simple commant into outgoing |
| 15:23.07 | brpvieira | havokmon: that's inherited from the freepbx dialplan, you could do it manualy with entries to the custom diaplan but it might be safer, and easier when you upgrade to add rules via the web interface |
| 15:23.56 | davevg-btwtech | davidnicol, did you reload the dialplan after making the change? |
| 15:24.12 | ManxPower | Havokmon: then perhaps you should be on the TRIXBOX CHANNEL |
| 15:24.18 | davidnicol | i did, with "reload" in the CLI |
| 15:24.37 | *** join/#asterisk stefmtl (n=root@stef.istop.com) |
| 15:25.32 | davidnicol | it could be my command file, it just had Channel and Extension in it -- it does dial the Channel |
| 15:25.46 | CapRicORN^80 | please check this : http://pastebin.com/m62a36a33 |
| 15:26.10 | ronr_ | anyone knows anything about point to point setting in asterisk? (for a PRI ISDN)? |
| 15:26.37 | ManxPower | davidnicol: .call files must be a specific format and syntax. |
| 15:27.01 | ManxPower | ronr_: PRIs do not have a point-to-point mode, only BRIs do |
| 15:27.04 | Qwell | ManxPower: echo "plzmakecall\nkthx" > myfile.call |
| 15:27.15 | davidnicol | ManxPower: what is the minimum? |
| 15:27.42 | stefmtl | I have errors when compiling svn-zaptel on Ubuntu /usr/src/zaptel/tor2.c:603: error: impossible constraint in asm. Which dependency do I miss ? |
| 15:27.54 | ManxPower | davidnicol: chances are you can figure that out from the sample.call file that is included in the Asterisk source, as well as the mailing list archives and voip-info.org |
| 15:28.08 | ronr_ | ManxPower: I expected that, still my telco tells me I need to configure the PBX point to point, not point to multipoint :S |
| 15:28.29 | ManxPower | ronr_: ignore them |
| 15:28.32 | Havokmon | ManxPower, rawr hiss :P I thought it was an asterisk macro :) |
| 15:29.06 | ronr_ | ManxPower: yeah... but I still don't have a signal on the dchan :( they say everything is ok and just tested on their side |
| 15:29.09 | ManxPower | Havokmon: The last person that asked a trixbox question here was fed to the 'gators. |
| 15:29.19 | Havokmon | yeah I've seen that ;) |
| 15:29.23 | CapRicORN^80 | anyone can help me |
| 15:29.27 | Qwell | which means the pigs are still hungry |
| 15:29.46 | Havokmon | It's just that there's no obvious line to where one begins and one ends |
| 15:29.52 | Qwell | Havokmon: exactly |
| 15:30.00 | Qwell | because they break everything |
| 15:30.11 | Qwell | which is why we can't (won't) support it here |
| 15:30.14 | ManxPower | define "signal on d-channel" |
| 15:31.16 | ronr_ | ManxPower: If I call out, I get a congestion error, pri debug shows no communication, and stuff like No D-channels available! Using Primary channel 16 as D-channel anyway |
| 15:31.17 | ManxPower | I assume you mean "no D-Channel messages when I do a pri debug span x", but for all I know you could be getting a Red or Yellow alarm and thinking that is "no signal on d-channel" |
| 15:31.35 | ronr_ | ManxPower: there are no alarm on the card |
| 15:31.36 | Havokmon | That's fine, I'm just saying it may not always be obvious where the ownership is |
| 15:31.43 | ManxPower | ronr_: then you tell your telco "get your ass down here with a T-BERD to test the D-channel". |
| 15:31.47 | tzafrir_laptop | stefmtl, what version of Ubuntu? What svn branch of Zaptel? |
| 15:32.01 | ronr_ | ManxPower: :D |
| 15:32.24 | ManxPower | ronr_: I'm not joking. If you have no traffic on the D-channel, then you have a major issue with your line. |
| 15:33.00 | ManxPower | and the "no red alarm, no traffic on the d-channel" almost always means the telco will say it is your fault until they have a tech on-site with something to test the PRI with. |
| 15:33.17 | ronr_ | ManxPower: I'm calling them again now, they said they testing everything and all is ok |
| 15:33.31 | brpvieira | ronr: have you checked the switches on your e1 modem? |
| 15:33.34 | stefmtl | tzafrir_laptop : Ubuntu 7.10 Server. Zaptel SVN-trunk-r3304 |
| 15:33.47 | ManxPower | ronr_: I'll bet they are testing the T-1 and you are not having problems with the T-1, just the D-channel |
| 15:33.55 | tzafrir_laptop | stefmtl, use branches/1.4 |
| 15:34.00 | ronr_ | brpvieira: what switches? |
| 15:34.03 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 15:34.12 | tzafrir_laptop | zaptel trunk is not used. svn help switch |
| 15:34.15 | stefmtl | tzafrir_laptop : ok thanks |
| 15:34.24 | brpvieira | the local/remote loop test switches... |
| 15:34.46 | ManxPower | ronr_: think of the T-1 as an Ethernet connection. You have a link light (no red alarm) or you don't have a link light (red alarm), but that does not mean you can connect to hosts, there are higher level protocals that have to be working as well. |
| 15:34.49 | ronr_ | brpvieira: no, how would I do that (I got a serial connection to the modem) |
| 15:34.54 | brpvieira | I dunno all T1/E1 cpes have these but at least all I've seen do |
| 15:35.10 | ronr_ | brpvieira: I'll check |
| 15:35.36 | tzafrir_laptop | stefmtl, though I suspect 'svn co http://svn.digium.com/svn/zaptel/branches/1.4 zaptel-1.4' would be shorter, with the volume of changes... |
| 15:35.54 | brpvieira | normally the telco will tell you to open/close the loops to test the line, maybe someone left them in loop after the test |
| 15:35.59 | ronr_ | brpvieira: could that be a loop option on the modem in ntp/e1 with options off / local and remote |
| 15:36.01 | *** join/#asterisk variable_office (n=variable@cerberus.iswan.net) |
| 15:36.19 | ManxPower | ronr_: What "modem"???? |
| 15:36.26 | variable_office | i am having a strange problem, both zaptel and asterisk are not making any init.d scripts; any ideas? |
| 15:36.49 | ronr_ | ManxPower: Athrea AC 2032/t |
| 15:37.04 | tzafrir_laptop | variable_office, "making" == ? Also: what distribution is it? What versions? |
| 15:37.08 | ManxPower | ronr_: Why is a modem on your T-1/E-1? |
| 15:37.30 | stefmtl | tzafrir_laptop : the svn version is not working properly ? |
| 15:37.37 | variable_office | newest, 1.4.16.2 i beleive; just downloaded this morning; on centos 4.6 which is essentially rhel |
| 15:37.39 | ronr_ | ManxPower: that's the box the telco installed |
| 15:37.46 | ManxPower | Telco ===> smartjack ===> Sangoma or Digium Card |
| 15:37.57 | ManxPower | ronr_: that's called a smartjack in most places. |
| 15:37.57 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 15:37.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:38.01 | variable_office | tzafrir_home, yes, i am compiling from source, and the init.d scripts are not being installed |
| 15:38.01 | ronr_ | it has a port marked E1 that I connected to the digium card |
| 15:38.13 | ManxPower | variable_office: they are not installed by default. |
| 15:38.19 | ronr_ | ok, so I should just try to remove the mode |
| 15:38.36 | variable_office | ManxPower, they always used to be, i was running 1.4.14 and they were installed by default |
| 15:38.39 | ronr_ | *modem |
| 15:38.45 | *** join/#asterisk beek (n=klinebl@65.211.106.243) |
| 15:38.45 | ManxPower | ronr_: I strongly doubt that box is a modem. |
| 15:39.20 | ManxPower | variable_office: no they were not. They are installed by a special make command. |
| 15:39.32 | CapRicORN^80 | why i getting message "Asterisk died with code 1 after giving command safe_asterisk |
| 15:39.35 | variable_office | what is the special make command? |
| 15:39.39 | ManxPower | I would have to look up the command. |
| 15:39.51 | ManxPower | CapRicORN^80: that means "asterisk did not start" |
| 15:40.11 | brpvieira | ManxPower: they call this sort of cpe a modem here, maybe mistranslation |
| 15:40.27 | CapRicORN^80 | then what should i do ? |
| 15:40.35 | ronr_ | ManxPower: this is the box: http://www.aethra.com/ProductDetails.asp?dp=84_AC,2032 |
| 15:40.43 | ManxPower | "make config" is the command to install the init scripts |
| 15:41.43 | ManxPower | ronr_: Do you see the word "modem" ANYHWERE on that page? |
| 15:41.56 | variable_office | I know for a fact that 1.4.12 put the init scripts in the /etc/init.d/ directory for me upon make, make install, or make samples because it was created and I am using the directions I wrote when installing 1.4.12 now |
| 15:42.15 | ronr_ | ManxPower: you're right, there isn't |
| 15:42.28 | ManxPower | ronr_: that is what we call a "smartjack", it converts the line from whatever the telco used to deliver it into a standard DSX-1 interface for customer equipment. |
| 15:42.43 | ronr_ | ManxPower: ah, ok |
| 15:43.02 | ManxPower | The telco can use SDSL (the most common) fiber, coax, or carrier pidgen to get the line to you, then that box converts the line into what you ordered. |
| 15:43.32 | ronr_ | so I gotta call the telco again and tell them the point to point option simply doesn't exist on the PRI only on BRI |
| 15:43.43 | ManxPower | I believe "smartjack" is a specific make/model of device. Much like the word "xerox" and "kleenex" it has become a more generic term. |
| 15:44.01 | ManxPower | ronr_: no, call them up and tell them you set the option and it still does not work |
| 15:44.09 | ronr_ | ok |
| 15:44.15 | ManxPower | sometimes you do have to lie. |
| 15:44.50 | ManxPower | ronr_: I suspect the E-1 is installed and working correctly, but the higher level protocol (D-channel) is not set up correctly in their switch. |
| 15:44.55 | *** join/#asterisk codn (n=codaine@85.107.249.50) |
| 15:45.18 | ManxPower | and the only way to get them to admit that is for a tech to plug into the smartjack with a device that supports PRI (D-channel) testing. |
| 15:45.44 | ManxPower | force the tech to make a call using the test box, if he can't make a test call, the line is not working. |
| 15:45.47 | ronr_ | ManxPower: ok |
| 15:46.07 | ManxPower | they may send a tech out with a box that ONLY tests the E-1, not the PRI (Q.931) protocols. |
| 15:46.21 | ManxPower | Make the tech try to call out the PRI using their test box. |
| 15:46.37 | ronr_ | I will |
| 15:46.57 | ManxPower | About 1/3 to 1/4 of the PRIs I've had to connect to Asterisk had exactly this problem when they were installed. |
| 15:47.16 | ManxPower | Only once was the problem actually Asterisk. |
| 15:48.03 | ManxPower | (BTW, the problem in Asterisk was a mismatch between Asterisk and Zaptel versions) |
| 15:48.17 | *** join/#asterisk tobias (n=tobias@nat1.ppckernel.org) |
| 15:49.45 | ronr_ | ManxPower: everything was downloaded latest version 2 days ago, so that should be ok |
| 15:50.00 | *** join/#asterisk Bog_ (n=Bog_@surf99.net.rss.rogers.com) |
| 15:50.07 | Bog_ | hello |
| 15:50.26 | [gnubie] | hello again.. |
| 15:50.49 | Bog_ | I've herd alot about Astrisk and curious to know how others use it at home or in a small home office |
| 15:51.16 | [gnubie] | can you tell me what are the disadvantages of using pattern matching in the dialplan? |
| 15:51.17 | Bog_ | I'm planning some home automation projects and curious to know what I might be able to do with Asterisk |
| 15:51.35 | ManxPower | Bog_: You should not think of Asterisk as a PBX or a phone system. Think of Asterisk as a TOOLKIT that allows YOU to build a PBX aka phone system. It's not a turnkey product. |
| 15:51.37 | Bog_ | gnu: no idea |
| 15:51.58 | cpm | Bog_, cool concept. |
| 15:52.03 | Bog_ | ManxPower: ok, I'm just thinking of some applications for use with a home automation system |
| 15:52.19 | ronr_ | those telco guys really don't know shit... |
| 15:52.40 | ManxPower | ronr_: all the ones that do end up quitting or going into management. |
| 15:52.47 | [gnubie] | Bog_: i heard people implemented asterisk and mythtv.. you can also use asterisk as your alarm clock, check the weather, etc.. |
| 15:53.28 | Havokmon | ronr_, I had a Fujitsu 9600 guy tell me once any modems that went through the Fujitsu would be limited to 9600 baud... *rolls eyes* |
| 15:53.43 | Bog_ | gnu: ok.... I'm planning to use a system called Control4 to control my home... distribute audio, control blinds, control lights and control themostat |
| 15:54.01 | Alexandre_fr | bog_ you can use it to select number, if the number is known in you database: all your phones ring, otherwise -> answering machine |
| 15:54.05 | [gnubie] | Bog_: good luck! :) |
| 15:54.45 | Bog_ | gnu: I have it all designed, but just curious about what I may do with Asterisk in the mix as well |
| 15:55.53 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:55.53 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 15:56.12 | Bog_ | gnu: I suppose I could try and create a gateway to control the home by calling asterisk |
| 15:56.50 | [gnubie] | can anyone tell me the possible disadvantages of using pattern matching over an explicit listing of extensions and apps in the dialplan? |
| 15:57.19 | variable_office | does anyone know the command to install the asterisk and zaptel init.d scripts? |
| 15:57.30 | Alexandre_fr | gnubie, i don't think there is disavdvantages |
| 15:59.43 | Alexandre_fr | asterisk always go to the more explicit extensions |
| 15:59.44 | beek | variable_office: I think it's make config |
| 16:00.03 | Alexandre_fr | if you have exten => 111 |
| 16:00.19 | Alexandre_fr | and exten => _1XX |
| 16:00.34 | [gnubie] | Alexandre_fr: do you use extensively pattern matching on your dialplan? |
| 16:00.38 | *** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 16:00.48 | Alexandre_fr | if you call 111 it will go to 111 |
| 16:01.32 | Bog_ | what kind of IP phones can be used with Asterisk? |
| 16:01.44 | Alexandre_fr | gnubie not a lot |
| 16:01.47 | *** join/#asterisk data23 (i=data@92.b6.3845.static.theplanet.com) |
| 16:01.50 | [gnubie] | Bog_: there's so many of them |
| 16:02.02 | [gnubie] | Alexandre_fr: what about the others? |
| 16:02.17 | Bog_ | gnu: I'm looking for some on the website |
| 16:02.39 | [gnubie] | Bog_: snom, grandstream, polycom, etc.. |
| 16:02.45 | *** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-c531125a13e2735c) |
| 16:03.00 | [gnubie] | Bog_: http://www.snom.com/ |
| 16:03.06 | Bog_ | thanks |
| 16:03.09 | [gnubie] | Bog_: http://www.grandstream.com/ |
| 16:03.32 | [gnubie] | Bog_: or, get a Nokia eSeries mobile phone |
| 16:03.56 | [gnubie] | Bog_: the nokia e51 is nice, light and thin.. :) |
| 16:04.03 | Bog_ | cool |
| 16:04.17 | Alexandre_fr | gnubie: http://www.asteriskdocs.org/ |
| 16:04.22 | Alexandre_fr | dl the book |
| 16:04.38 | *** join/#asterisk ddunavant (n=David@66.170.97.28) |
| 16:04.46 | [gnubie] | Alexandre_fr: actually, i already bought it but haven't read it yet.. :( |
| 16:04.52 | Alexandre_fr | go page 137 |
| 16:05.04 | Alexandre_fr | there is something about pattern |
| 16:05.06 | [gnubie] | wait.. i'll check it.. |
| 16:06.59 | Alexandre_fr | I have to go good luck with asterisk :!) |
| 16:07.03 | *** part/#asterisk Alexandre_fr (n=alex@nat04-bzr.dys1.com) |
| 16:12.33 | *** join/#asterisk CVirus (n=GoD@196.205.192.155) |
| 16:14.45 | hi365 | ManxPower: i just noticed core.xxx files in my system tmp. will these be helpful to debug asterisk? |
| 16:15.19 | hi365 | there like 100 mb each... |
| 16:15.28 | hi365 | there like 100 mb each... |
| 16:15.45 | hi365 | ~asterisk |
| 16:15.46 | jbot | i guess asterisk is the best free PBX in the world, or #asterisk on irc.freenode.net, or http://www.asterisk.org |
| 16:16.18 | *** join/#asterisk hi365 (n=hi365@mail.pcgeula.co.il) |
| 16:16.26 | *** part/#asterisk CVirus (n=GoD@196.205.192.155) |
| 16:17.49 | *** join/#asterisk shadebob (n=chatzill@84.16.28.38) |
| 16:17.51 | shadebob | hi |
| 16:23.36 | *** part/#asterisk harpal (n=Harpal@124.125.79.212) |
| 16:26.18 | tzafrir_laptop | hi365, generally: gdb -c /path/to/core.file /usr/sbin/asterisk |
| 16:26.22 | tzafrir_laptop | bt |
| 16:26.25 | tzafrir_laptop | bt full |
| 16:27.01 | tzafrir_laptop | # some keywords for search: asterisk gdb backtrace core crash |
| 16:27.01 | hi365 | tzafrir_laptop: thanks. i just noticed some core.xxxx files in /tmp.can these be useful? |
| 16:27.14 | tzafrir_laptop | sure. See above |
| 16:27.31 | hi365 | ah. thought that was just to start it. gona have a look |
| 16:29.11 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 16:31.33 | *** join/#asterisk LucanUK (i=aolsux@77-100-10-182.cable.ubr06.harb.blueyonder.co.uk) |
| 16:34.41 | LucanUK | hey everyone :) sort of hitting my head against a brick wall and hoped someone might be able to help me..... |
| 16:34.49 | LucanUK | so does anyone have any knowledge of chan_mobile? when im doing a "mobile search" the port numbers change after every module reload |
| 16:35.29 | CapRicORN^80 | i am trying to call top my other softphone (xlite) when i dial it say called: declined |
| 16:35.56 | CapRicORN^80 | i hear the tape "the person you are calling is unavaiable " |
| 16:36.22 | CapRicORN^80 | i have stopped my iptables |
| 16:36.30 | CapRicORN^80 | what may be the cause of it ? |
| 16:39.03 | CapRicORN^80 | hello ? |
| 16:39.24 | Kobaz | sdflkhsjdfsdf |
| 16:39.28 | Kobaz | this polycom phone sucks ass |
| 16:39.42 | hi365 | tzafrir_laptop: does this mean anything? Program terminated with signal 11, Segmentation fault. #0 0x0017c30b in strlen () from /lib/libc.so.6 |
| 16:39.54 | Kobaz | hi365: means something bad happened, file a bug report |
| 16:40.21 | hi365 | Kobaz: is that all the info i will need to post? |
| 16:40.22 | Kobaz | turn on core dumping, do a backtrace on it, and send it off to the team |
| 16:40.40 | Kobaz | you'll need more, that by itself is fairly useless |
| 16:43.08 | russellb | hi365: paste the whole backtrace to pastebin |
| 16:43.10 | codefreeze | hi365: what Kobaz means is when you configure asterisk, do a make menuselect, and mark the "compiler options" entry for "no optimize". then compile, and run under gdb. in gdb do a "thread apply all bt" and attach that output to your bug report. |
| 16:43.13 | russellb | gdb /usr/sbin/asterisk core.12345 |
| 16:43.20 | russellb | (gdb) bt ... (gdb) bt full |
| 16:43.57 | af_ | the sound adapter is needed for music on hold? |
| 16:45.17 | mvanbaak | af_: no |
| 16:45.57 | af_ | so no more mpg123? |
| 16:46.18 | af_ | asterisk uses anymore the sound adapter at all? |
| 16:46.28 | af_ | (apart chan_oss) |
| 16:46.37 | mvanbaak | chan_alsa |
| 16:47.41 | af_ | right. so anymore else uses sound adapter? |
| 16:50.04 | hi365 | thanks all. ill try to get this set up. im having this problem on two production machines - one with the lates everything (libpri, zaptel, asterisk, wanpipe) |
| 16:50.48 | hi365 | codefreeze: do i need to recompile everything (libpri, zaptel asterisk, aa) or can i just recompile asterisk |
| 16:52.53 | tzafrir_laptop | Just Asterisk |
| 16:53.22 | hi365 | thanks |
| 16:54.02 | mvanbaak | af_: no |
| 16:56.07 | *** join/#asterisk Schumie (i=SteveWri@cpc1-rdng2-0-0-cust233.winn.cable.ntl.com) |
| 16:56.54 | *** join/#asterisk nybble (n=nybble@about/apple/performa/nybble) |
| 16:58.13 | Bog_ | what VoIP providers would you recommend that support sterisk? |
| 16:58.27 | Qwell | ~itsplist |
| 16:58.32 | Qwell | stupid bot |
| 16:58.40 | Qwell | ~itsp |
| 16:58.40 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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 |
| 16:59.11 | Qwell | Bog_: see above - pick a list |
| 16:59.12 | Bog_ | ~itsplist-ca |
| 16:59.13 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 16:59.25 | Bog_ | thanks |
| 16:59.31 | Qwell | babytel? that sounds so wrong |
| 16:59.43 | russellb | jbot: thwack jbot |
| 16:59.43 | jbot | ACTION bludgeons jbot on the head with a 5ESS Switch |
| 16:59.54 | Qwell | telephony related thwacking++ |
| 17:00.35 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583811.dsl.bell.ca) |
| 17:01.14 | ManxPower | "Our packets are made out of 100% babies! No fillers!" |
| 17:01.26 | *** join/#asterisk Winkie (n=urmom@149.254.192.192) |
| 17:02.53 | Qwell | ManxPower: that's slightly ambiguous. is 1 packet 100% of a baby? |
| 17:03.23 | ManxPower | Qwell: It's marketing. It's supposed to be ambiguous. 8-) |
| 17:03.27 | Qwell | ;) |
| 17:04.11 | russellb | i would hate to be a call on their network ... delivering a baby every 20 ms ... sounds painful |
| 17:04.35 | ManxPower | You need to use them for SOMETHING. |
| 17:05.40 | JunK-Y | and thats not unlimitel.com, its unlimitel.ca :) |
| 17:08.12 | hi365 | so once i recompiled asterisk with DONT_OPTIMIZE i just sit back and wait for the thing to crash? |
| 17:09.17 | ManxPower | hi365: welcome to Asterisk. Most of the time the fix to a bug is easy. Sometimes it's not. |
| 17:09.41 | ManxPower | s/Asterisk/life |
| 17:10.12 | hi365 | is there anyway i can confirm that tis is indeed runing with not optimized? |
| 17:15.32 | RoyK | hi365: you'll see it from the crashdump. dunno if there are ways to see it from the running code, though |
| 17:16.18 | RoyK | hi365: also, while compiling, gcc should not run with -O2 or -O(anything) except perhaps -O0 |
| 17:16.18 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 17:16.20 | hi365 | hmm, ill recompile my other system and ill pay attension then |
| 17:16.35 | *** join/#asterisk iratik (n=itariki@12-226-116-3.client.mchsi.com) |
| 17:16.45 | RoyK | hi365: make clean && make all > make.log 2>&1 && make install |
| 17:17.05 | dezenten | what makes the red light go on my linksys spa941 when i get a voicemail ? is ia sip-message ? |
| 17:17.11 | iratik | letse say a call comes into a certain extension 512, is there any way for extension 1518 to take over 512's call ? |
| 17:17.30 | hi365 | whats make all? i never used that one before |
| 17:17.52 | *** join/#asterisk angeldavid (n=pepOSX@nelug/coreteam/pepo) |
| 17:18.00 | hi365 | iratik: check out Pickup() |
| 17:18.01 | RoyK | hi365: just make install, then. make all is implicit in make install |
| 17:18.15 | RoyK | make all compiles it all - make install installs the binaries |
| 17:18.20 | iratik | hmmm |
| 17:18.21 | iratik | thanks |
| 17:19.24 | *** join/#asterisk codn (n=codaine@88.240.59.153) |
| 17:20.10 | iratik | hi365: is that "Pickup()" this ?http://www.voip-info.org/wiki-PBX+Call+Pickup ? |
| 17:21.05 | hi365 | i actualy meant this: http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup |
| 17:22.32 | codefreeze | hi365: all you need to recompile is just asterisk, sorry for the delay, btw! |
| 17:23.06 | codefreeze | oops, looks like someone answered it |
| 17:23.44 | hi365 | codefreeze: if only more people here would be that anxious to help others! |
| 17:25.18 | jbot | shoves a crumpet down codefreeze :)'s throat, happy now?! Huh? Want some JAM with that?, courtesy of royk |
| 17:26.28 | *** join/#asterisk phalacee (n=phalacee@123-2-59-211.static.dsl.dodo.com.au) |
| 17:26.51 | codefreeze | RoyK: how did you do that? |
| 17:27.11 | RoyK | /msg jbot lart #asterisk codefreeze |
| 17:28.05 | codefreeze | RoyK: aha... many thanks, both for the lart and the lart method. (would this be lartology?) |
| 17:28.27 | RoyK | ~lart codefreeze |
| 17:28.27 | jbot | takes out a seltzer bottle and sprays codefreeze in the face. You know, one of those old-school seltzer bottles clowns have? Yeah those. Anyway, consider yourself spritzed |
| 17:28.33 | RoyK | :) |
| 17:30.58 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:38.44 | *** join/#asterisk amir_ (n=amir@gentoo/developer/amir) |
| 17:52.04 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 17:52.05 | *** mode/#asterisk [+o anthm] by ChanServ |
| 17:55.31 | *** join/#asterisk Skarmeth (n=Skarmeth@201.9.74.32) |
| 18:01.47 | *** join/#asterisk ZaVoid (n=zavoid@c-68-44-91-5.hsd1.nj.comcast.net) |
| 18:01.51 | ZaVoid | hi all |
| 18:01.56 | ZaVoid | everyone have good holiday? |
| 18:07.34 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:15.22 | syzygyBSD | I have now that you are here |
| 18:16.00 | Qwell | ZaVoid: it was alright, but I bet if somebody *cough*cough* were to give me a million dollars, it'd be pretty good |
| 18:16.09 | ZaVoid | same here Qwell |
| 18:16.14 | ZaVoid | still waiting for that angel funding eh? |
| 18:16.46 | *** join/#asterisk ddunavant (n=David@66.170.97.28) |
| 18:17.46 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:18.22 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 18:18.44 | [T]ank | I started seeing an error and cannot figure out what it means. Could anyone assist? ast_readaudio_callback: Failed to write frame |
| 18:22.17 | ManxPower | [T]ank: That's just fancy talk for "call disconnected". It does not indicate a problem. |
| 18:22.34 | [T]ank | ManxPower: ok, thanks |
| 18:22.43 | *** join/#asterisk Winkie (n=urmom@149.254.192.192) |
| 18:23.19 | *** join/#asterisk _ShrikE (n=ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 18:24.02 | ZaVoid | anyone ever try one of these SigC5561 cards? |
| 18:24.03 | ZaVoid | http://www.signalogic.com/index.pl?page=asterisk_ip_pbx |
| 18:24.34 | hi365 | anyone have any syntax highliting for gedit? |
| 18:24.36 | ZaVoid | sounds like it can do a ton more transcoding then the tcb400 junk |
| 18:24.53 | *** join/#asterisk CVirus (n=GoD@196.205.192.155) |
| 18:25.09 | ManxPower | ZaVoid: Every card I've seen has not been cheaper enough to move away from Digium or Sangoma |
| 18:25.38 | ManxPower | Not familiar with that specific card, however. Also remember if not many people use the card you're screwed if you need help. |
| 18:25.56 | ZaVoid | yeah i know |
| 18:26.21 | ZaVoid | i'm using a Cantata IMG1010 for transcoding right now because the tcb400's from digium didn't work right with g.723 flavor negotiation |
| 18:26.38 | ZaVoid | but at 50k a pop for a IMG1010 i'm always looking for better alternatives |
| 18:27.01 | Qwell | ZaVoid: iirc, the signalogic doesn't actually...exist |
| 18:27.16 | ZaVoid | speaking of digium... has anyone tried out that IAX loadbalancer that gradwell released a few months ago? |
| 18:27.17 | Qwell | and there were fixes recently to g723 on the tc400 |
| 18:27.17 | ManxPower | Also exactly how much does the card cost? |
| 18:27.55 | ZaVoid | yeah but we bought the card in june(ish) and spent two months fighting with digium to figure out what was wrong and then try to get it worked and when that failed we had to go a carrier grade solution |
| 18:28.19 | ZaVoid | Qwell: just marketing fluff? that makes sesne.. the channel density for transcoding on one card seems really high |
| 18:28.27 | ZaVoid | ManxPower: i can't find a price for it |
| 18:28.57 | ZaVoid | i was thinking of testing the gradwell iax load balancer unless anyone else has another recomendAtion to load balance iax traffic? |
| 18:28.58 | ManxPower | ZaVoid: There have been several cards from various companies that were announced, but NEVER shipped. |
| 18:30.31 | ZaVoid | http://cantata.com/products/img1010/ |
| 18:30.33 | ZaVoid | nice product |
| 18:31.15 | ManxPower | For 50k it had better buy me dinner AND put out. |
| 18:31.18 | ZaVoid | manx/qwell you guys every play with gradwell's iax load balancer? |
| 18:31.25 | ZaVoid | they bought drinks.. that worked |
| 18:31.34 | ManxPower | ZaVoid: We don't use IAX |
| 18:31.38 | ManxPower | At least anymore. |
| 18:31.53 | *** join/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 18:31.58 | ZaVoid | their box is way better for ss7 then cisco's mgc solution too |
| 18:31.59 | ZaVoid | really? |
| 18:32.02 | ZaVoid | your lucky |
| 18:32.09 | ZaVoid | my expereince with it has been nothing but horrible |
| 18:32.15 | ZaVoid | only part of my customer network still on 1.2 |
| 18:32.17 | ManxPower | Switched to all SIP and all our problems went away. |
| 18:32.34 | ZaVoid | any 1.4 train i try fails miserably and crashes(slow non core dumping kinda crash) |
| 18:32.44 | ZaVoid | some of our customers only use iax though :( |
| 18:33.07 | russellb | if you have problems, _please please please_ report them, so i can fix them |
| 18:33.11 | ZaVoid | hi russellb |
| 18:33.12 | russellb | i have fixed a ton of iax2 problems in the 1.4 series |
| 18:33.24 | ZaVoid | last 1.4 we tried for iax was 1.4.4 i think |
| 18:33.26 | russellb | it should work great in the latest versions of 1.4. |
| 18:33.28 | russellb | oh good lord |
| 18:33.32 | ZaVoid | old i know |
| 18:33.34 | variable_office | on a zaptel card, should the phone have to ring once before asterisk detects it? (fxo card) |
| 18:33.40 | russellb | all of the fixes i have made have been after that |
| 18:33.40 | ZaVoid | but we've been putting it off for a while |
| 18:33.50 | ZaVoid | mind if i pm you for one sec russelb? |
| 18:33.54 | russellb | yes :) |
| 18:34.00 | ZaVoid | oh :( |
| 18:34.06 | ManxPower | ZaVoid: IAX2 was basically totally broken for any kind of higher than occasional calls in all 1.4 releases until recently. |
| 18:34.08 | russellb | unless you're sure a pm is necessary ... |
| 18:34.11 | russellb | i'd rather just talk in here. |
| 18:34.23 | ZaVoid | just 2 please :) |
| 18:34.25 | ZaVoid | then back here |
| 18:34.35 | ManxPower | ZaVoid: It sounds like you are a service provider. why are you not on the asterisk-dev or asterisk-cvs mailing lists? |
| 18:34.42 | russellb | if you must ... |
| 18:34.43 | ZaVoid | i do read them |
| 18:35.05 | ManxPower | You missed the discussions of the massive fixes that were put into 1.4 for IAX2? |
| 18:35.21 | jer | holy shit, i finally got authorization to update our asterisk implementation from 1.2 to 1.4 |
| 18:35.28 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 18:35.36 | ManxPower | jer: You poor, poor thing. |
| 18:35.48 | jer | ManxPower, i've been fighting for months =/ it's still on 1.2.17 |
| 18:36.02 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583811.dsl.bell.ca) |
| 18:36.21 | ManxPower | jer: If you don't read both the 1.2 upgrade.txt and the 1.4 upgrade.txt you are in for a rough ride. |
| 18:36.36 | jer | ManxPower, i know |
| 18:37.06 | ZaVoid | russellb: we also found that for SIP 1.4.9 and 1.4.15 were the only stable builds we cold find... anything in between those crashed on average 3-4 times a day(core dumps and no i didn't submit them sorry) |
| 18:37.08 | ManxPower | jer: be glad it took so long to get permission. 1.4 seems like it's stablized in recent releases. I've not used it, but I've seen far fewer reports of major issues with 1.4 |
| 18:37.23 | harryv | i have a register => 1234:passwd@host.tld/1234 in sip.conf.. shouldnt `exten => ._,1,Blabla` catch that one? |
| 18:37.56 | ManxPower | ZaVoid: In my experience it's pretty useless to submit bug reports for problems on a production server. You just can't have the downtime required to troubleshoot the problem on production servers. |
| 18:38.00 | russellb | ZaVoid: i understand, that's what most people do ... find something that works and use it |
| 18:38.11 | ManxPower | harryv: you never want to use _. |
| 18:38.20 | harryv | _X. then |
| 18:38.28 | ZaVoid | yeah we called 1.4.9 the magic build |
| 18:38.36 | ZaVoid | but then 1.4.9 didn't have the working rtp cut off |
| 18:38.37 | ManxPower | that would work, assuming your provider is actually sending a destination number with the call. |
| 18:38.39 | ZaVoid | so we found 1.4.15 |
| 18:38.44 | *** join/#asterisk fukz (n=basti@p5B062132.dip.t-dialin.net) |
| 18:38.45 | ZaVoid | and we use realtime and pgsql too |
| 18:38.51 | jer | ManxPower, yeah |
| 18:39.31 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 18:39.49 | ManxPower | At this point even I would consider using 1.4 |
| 18:40.06 | ZaVoid | 1.4.9 is pretty stable manx.. |
| 18:40.11 | ManxPower | well, at least trying it. |
| 18:40.17 | ZaVoid | we ran at least 3 months on it on 8 boxes with 0 crashes |
| 18:40.22 | ZaVoid | we are stil shocked about it |
| 18:40.26 | ManxPower | ZaVoid: Stable for YOUR usage patterns you mean. |
| 18:40.27 | ZaVoid | 1.4.15 seems to be the same. |
| 18:40.29 | ZaVoid | yes |
| 18:40.46 | ZaVoid | but i'm pumping about 30-150 concurrent sip calls over it no transcoding on the boxes though |
| 18:40.54 | *** part/#asterisk CVirus (n=GoD@196.205.192.155) |
| 18:41.10 | ManxPower | The usage patterns of my customers is a corporate environment, not an ITSP environment. |
| 18:41.31 | *** join/#asterisk Downchuck (n=chatzill@c-98-203-140-155.hsd1.wa.comcast.net) |
| 18:41.52 | ManxPower | ITSP's customers seem to be able to handle more downtime, than the prima donna users in a corporate enviroment. |
| 18:43.28 | ZaVoid | ah |
| 18:43.33 | ZaVoid | more pbx phones then wholesale? |
| 18:44.10 | Bog_ | what asterisk appliances are there? |
| 18:44.21 | Qwell | Bog_: there is the s800i from Digium |
| 18:44.35 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 18:44.39 | Bog_ | qwell: tht was the business license though |
| 18:45.35 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 18:45.51 | *** join/#asterisk implicit (i=implicit@gateway/tor/x-7ceb94ec4ff6936c) |
| 18:46.05 | Qwell | Bog_: perhaps you have a different definition of appliance then |
| 18:46.30 | Bog_ | qwell: I just want a hardware device with Asterisk installed and running |
| 18:46.52 | Qwell | sounds like the s800i fits that description |
| 18:47.01 | *** part/#asterisk fukz (n=basti@p5B062132.dip.t-dialin.net) |
| 18:47.20 | Bog_ | I only see aa50 |
| 18:47.22 | *** join/#asterisk techie (n=techie@adsl-76-214-31-16.dsl.lsan03.sbcglobal.net) |
| 18:47.25 | Qwell | Bog_: same thing |
| 18:47.28 | ZaVoid | so no one has tried that iax load balancer at all then? not even you russ? |
| 18:47.34 | Bog_ | ok |
| 18:48.07 | Bog_ | qwell: what editions of asterisk is there? |
| 18:48.26 | Qwell | there's open source, and BE (or in the case of the aa50, Embedded BE) |
| 18:48.36 | Bog_ | qwell: this one says business edition with commercial licenses... I want an oss version |
| 18:49.13 | ManxPower | ZaVoid: We use exclusively PRIs, Polycom, Sangoma, Cisco. Hell, my users don't even PAY for calls. |
| 18:49.23 | ManxPower | Bog_: you will not get an OSS appliance. |
| 18:49.28 | ZaVoid | gotcha ManxPower |
| 18:49.37 | Bog_ | ManxPower: ok |
| 18:49.52 | Qwell | Bog_: it kinda goes against the term "appliance" |
| 18:50.09 | Qwell | an appliance is something you shouldn't ever need to much with - including (and especially) rebuilding source |
| 18:50.09 | Bog_ | qwell: I dont think so |
| 18:50.13 | ManxPower | ZaVoid: the only VoIP we do is phone/asterisk, asterisk/qos/wan/asterisk and asterisk/failoverto/SIP ITSP. |
| 18:50.16 | Qwell | s/much/muck/ |
| 18:50.27 | Bog_ | qwell: it means no licensing additional software |
| 18:50.34 | *** join/#asterisk javb (n=javb@190.80.201.55) |
| 18:50.39 | Qwell | it isn't "additional" software.. it's asterisk |
| 18:50.43 | ZaVoid | gothcah Maxxed |
| 18:50.45 | ZaVoid | manx |
| 18:50.48 | ManxPower | Bog_: no, appliance means "the hood is welded shut" |
| 18:50.56 | Bog_ | but it suggests some of the costs is going towards the software |
| 18:51.02 | Qwell | Bog_: support |
| 18:51.12 | Bog_ | ok |
| 18:52.17 | javb | Asterisk is asking authentication for a caller based on its callerid.. cant understand why! (loook: http://dpaste.com/29127/) |
| 18:53.25 | ManxPower | javb: how do you know the device is not just sending it's callerid as it's userid? |
| 18:54.55 | ManxPower | In fact that message means exactly that the phone is sending that as it's userid and you do not have that userid configured in sip.conf |
| 18:55.48 | ManxPower | That message give NO information about what Asterisk is expecting for the SIP userid, other than the fact that it was not expecting "8099503041" as the userid |
| 18:57.00 | mocker | Anyone ever used McLeodUSA's SIP offering? |
| 18:57.21 | ZaVoid | nah |
| 18:57.36 | ZaVoid | javb: can you do a sip debug? |
| 19:00.29 | *** part/#asterisk myiagy (n=Jose@200.215.59.133) |
| 19:02.06 | ManxPower | apparently not. |
| 19:02.18 | *** join/#asterisk DarylVoip (n=daryl@c-71-224-42-97.hsd1.pa.comcast.net) |
| 19:07.15 | Havokmon | grrr. I can't figure out my PRI issue. I have a Mcleod PRI, priovisioned from an Adtran. I also have a Fujitsu 9600. I have an asterisk box with 2 T100Ps. Adtran-> fujitsu works. asterisk->fujitsu works. asterisk-> adtran doesn't work :(( |
| 19:07.52 | Havokmon | I even took my working card/cable from the fujitsu and plugged it into the adtran, and the PRI still won't come up. |
| 19:08.51 | Havokmon | there's no alarm, so there is a link, but asterisk always says down. Oddly enough, the Mcleod tech says the adtran always shows no link.. Not sure if that's true or if he's not reading it right... |
| 19:09.48 | Havokmon | hmm now my working span is down... grrr |
| 19:11.30 | *** join/#asterisk thansen|laptop (n=thansen@19.243.sfcn.org) |
| 19:11.30 | syzygyBSD | it isn't really working then is it? |
| 19:11.56 | davidnicol | no sound for playbacks. Placing a call from channel=mydeskphone@mycompany ; context: demo; extension: s, priority: 1 |
| 19:12.10 | davidnicol | works in that I get called and when I nswer it, the asterisk |
| 19:12.23 | davidnicol | CLI log tells me that it is trying to play back the demo files |
| 19:12.32 | syzygyBSD | davidnicol: do you have an exten => 1,1,Answer() anywhere |
| 19:12.54 | syzygyBSD | try to put in the answer before the playback |
| 19:13.07 | davidnicol | would that be bad if I did? I have the 1.4 default configs and have not added those. okay i will try that |
| 19:13.50 | davidnicol | yes, demo starts Wait, Answer, ... |
| 19:14.21 | syzygyBSD | are you going through any firewalls? |
| 19:14.31 | davidnicol | di i need to install a codecs package on Centos? |
| 19:14.43 | syzygyBSD | it shouldn't matter |
| 19:14.59 | davidnicol | i have opened all ports btn the machine I am devving on and the company *-1.2 |
| 19:15.15 | *** join/#asterisk p3 (n=liszt@CPE-65-26-242-251.wi.res.rr.com) |
| 19:15.17 | Havokmon | Ahh I think it bounced because I changed the clocking trying to get the asterisk->adtran to come up for even one moment |
| 19:15.49 | davidnicol | and I opened the RTP range as well |
| 19:15.52 | *** part/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 19:15.52 | Havokmon | I guess I can't really clock off the span that never comes up can I? :P |
| 19:16.25 | p3 | hey guys, i've got an old pentium 3 i'm throwing linux on, and all i want this tower to do is run asterisk, what distro would you guys recommend? i'm going b/w debian and slackware for it right now |
| 19:16.39 | syzygyBSD | davidnicol: all the ports.. tcp and udp? |
| 19:16.41 | russellb | debian!! |
| 19:16.47 | rob0 | either is fine, choose what you prefer |
| 19:17.00 | p3 | i'll go with debian then since i've got a disk handy |
| 19:17.24 | syzygyBSD | I like the whole "apt-get install asterisk" ease of getting everything up and running too |
| 19:17.24 | davidnicol | no udp only; I'll open tcp too. I thought RTP was all UDP. |
| 19:17.42 | p3 | yeah, apt is always nice, although i prefer bsd ports to apt |
| 19:18.26 | syzygyBSD | :) |
| 19:18.45 | *** join/#asterisk angom (n=angom@200.56.104.87.dsl.dyn.telnor.net) |
| 19:19.44 | ZaVoid | when i do a sip show channels i show about 72 active channels.. i know for a fact that maybe 20 of those are really active |
| 19:20.08 | davidnicol | opening tcp too did not do it |
| 19:20.11 | syzygyBSD | ZaVoid: there are two channels for every one that is up.. |
| 19:20.18 | ZaVoid | yeah i know that syzygyBSD |
| 19:20.22 | syzygyBSD | ie, there are two ends to the conversation... |
| 19:20.23 | davidnicol | brb |
| 19:20.30 | syzygyBSD | oh, nm then... |
| 19:20.33 | ZaVoid | lol |
| 19:20.55 | russellb | i personally prefer one sided conversations |
| 19:21.04 | russellb | they are much more enjoyable than actually talking to people |
| 19:21.25 | ZaVoid | lol but your a programmer.. thats not a surprise. |
| 19:21.53 | drmessano | people are overrated |
| 19:21.58 | russellb | most asterisk developers really hate phones, i think |
| 19:22.02 | russellb | or at least hate using them |
| 19:22.17 | DarylVoip | It's not just asterisk devs. It's all of us in telecom. |
| 19:22.24 | russellb | oh, ok, good |
| 19:22.31 | russellb | glad to know i'm not crazy |
| 19:22.38 | DarylVoip | I didn't say that. :) |
| 19:22.40 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 19:22.44 | russellb | ha .. |
| 19:22.46 | syzygyBSD | we connect other people so we don't have to talk to any of them |
| 19:22.50 | drmessano | lol |
| 19:22.54 | russellb | syzygyBSD: nice |
| 19:23.04 | rob0 | You don't have to be crazy to be using *, but it helps. |
| 19:23.10 | ZaVoid | daryl your not really in telecom we've told you this |
| 19:23.19 | ZaVoid | you don't have the right history |
| 19:23.29 | DarylVoip | That's true. But I'm honorary. |
| 19:23.35 | *** join/#asterisk cmdln (n=cmdln@adsl-76-208-120-230.dsl.tpkaks.sbcglobal.net) |
| 19:23.51 | ZaVoid | syzygyBSD: we connect families all over the world at extremly low costs so they have more money to pay taxes and provide roads to the mountains so we can go skiiing |
| 19:23.56 | ZaVoid | honorary works |
| 19:24.43 | syzygyBSD | I prefer gondolas over roads |
| 19:25.15 | ZaVoid | european? |
| 19:25.32 | rob0 | I've never seen that. I've seen gondolas going UNDER roads, but not over. |
| 19:26.01 | syzygyBSD | New Zealand |
| 19:26.04 | russellb | you people are all insane |
| 19:26.10 | syzygyBSD | hopefully, |
| 19:26.17 | ZaVoid | your a kiwi? |
| 19:26.19 | syzygyBSD | passport is getting back from london soo |
| 19:26.21 | syzygyBSD | n |
| 19:26.24 | ZaVoid | where a bouts? i spent 4 months there |
| 19:26.29 | ZaVoid | you guys brew some damn good beer |
| 19:26.41 | syzygyBSD | not a kiwi yet, couple more months of waiting |
| 19:26.53 | ZaVoid | drink montieths and speights beer when you get there |
| 19:26.56 | ZaVoid | you'll thank me |
| 19:27.06 | syzygyBSD | I spent 3 months there last year... |
| 19:27.07 | ZaVoid | where in nz you moving to? |
| 19:28.04 | syzygyBSD | I would like to end up in nelson, but there isn't a real market there for me. Probably end up in wellington and just take planes over to hike in nelson lakes and abel tasman |
| 19:28.28 | harryv | when i call into asterisk it's like the first sound file i play is being 'stumbled ' -- it doesnt always play the first second or so |
| 19:28.59 | syzygyBSD | harryv: put a wait(1) as your first exten |
| 19:29.15 | *** join/#asterisk toresbe (n=toresbe@33.80-203-102.nextgentel.com) |
| 19:29.17 | syzygyBSD | the issue is it starts playing before the other end gets the answer() |
| 19:30.25 | toresbe | Hey folks, I'm having some problems receiving an incoming call from an IAX trunk |
| 19:31.03 | toresbe | I've set it to context "demo" |
| 19:31.28 | toresbe | when I call my Asterisk server, I get: |
| 19:31.29 | toresbe | Dec 27 20:26:32 NOTICE[13807]: chan_iax2.c:6918 socket_read: Rejected connect attempt from 81.163.15.23, who was trying to reach 's@' |
| 19:31.47 | toresbe | any hints on what I should be looking at? |
| 19:32.31 | syzygyBSD | in iax2.conf what is the context= set to? |
| 19:32.38 | syzygyBSD | the first one in the file |
| 19:32.54 | syzygyBSD | sorry, iax.conf |
| 19:33.14 | harryv | syzygyBSD: that works if i call from my internal SIP-softphone. but calling from the outside it waits the specified time (i set it to 5 seconds for test), and playing 'hello-world' it goes something like this: '*5 seconds* *click* llo world" |
| 19:33.15 | ZaVoid | anyone ever deal with SIP blocking in UAE? |
| 19:33.18 | harryv | :/ |
| 19:33.54 | syzygyBSD | harryv: do you have an answer() on the exten? |
| 19:34.19 | toresbe | syzygyBSD: there is no context setting - apart from the one in my declaration |
| 19:34.29 | toresbe | in my declaration of the IAX link, that is |
| 19:34.48 | harryv | syzygyBSD: oh .. ....... thanks. |
| 19:34.50 | syzygyBSD | right, it appears the call is comming in unauthenticated |
| 19:35.20 | toresbe | syzygyBSD: unauthenticated? |
| 19:35.28 | toresbe | Oh, wait. When I define... hang on a sec please. |
| 19:37.43 | rob0 | ZaVoid, no, but I would try routing your SIP through openvpn. |
| 19:37.55 | ZaVoid | yeah did that |
| 19:37.57 | ZaVoid | no go |
| 19:38.06 | ZaVoid | seems to be filtering at isp level |
| 19:38.25 | toresbe | syzygyBSD: Hmm. |
| 19:38.30 | ZaVoid | what i mean is.. i put a router in vpn setup in country.. and had customers in uae route to the incountry vpn server |
| 19:38.37 | ZaVoid | and the invites never got to it |
| 19:39.15 | toresbe | syzygyBSD: I've registered to it, and now I have the username and secret in the [chaos] section as well. |
| 19:39.16 | syzygyBSD | you have to be able to monitor them phone calls |
| 19:39.39 | syzygyBSD | toresbe: one or the other, not both, |
| 19:39.42 | toresbe | should I omit the username and secret from that section? Am I by this saying that I expect that username and secret from the PBX? |
| 19:39.48 | rob0 | Put the router *outside* UAE, and local routers/clients VPN to that. Then SIP goes to the external router. |
| 19:39.50 | syzygyBSD | remove your register => line in iax.conf |
| 19:39.56 | toresbe | syzygyBSD: OK |
| 19:40.16 | toresbe | syzygyBSD: done |
| 19:40.37 | rob0 | Offer void if prohibited by law. :) |
| 19:40.39 | ZaVoid | clients have devices at locations but no where to put openvpn at same location |
| 19:40.42 | syzygyBSD | in the [chaos] section do you have the context = ? |
| 19:40.50 | toresbe | syzygyBSD: yes, context=demo |
| 19:41.02 | syzygyBSD | ok, try the call again, after a reload of course |
| 19:41.03 | toresbe | when I remove it from the register => line, asterisk doesn't notice when I call it anymore |
| 19:41.48 | syzygyBSD | toresbe: do you have the host= in the [chaos] |
| 19:41.55 | toresbe | syzygyBSD: yes |
| 19:42.08 | syzygyBSD | to an ip right, not dynamic? |
| 19:42.35 | syzygyBSD | in the asterisk console, what do you get when you do 'iax2 show registry' |
| 19:42.50 | syzygyBSD | oh, and is the type=friend in that context? |
| 19:42.51 | toresbe | no, to a hostname, not an IP |
| 19:43.17 | *** join/#asterisk asagage (i=asagage@12.192.197.15) |
| 19:43.22 | toresbe | now I get zero lines |
| 19:43.43 | syzygyBSD | this is all iax right? |
| 19:43.49 | toresbe | yes |
| 19:44.00 | toresbe | if I uncomment the register=> line, I get the line in the registyr |
| 19:44.09 | syzygyBSD | ok. can you sanitize and pastebin your iax.conf? |
| 19:44.36 | syzygyBSD | you shouldn't need the register => line |
| 19:45.23 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:45.29 | asagage | Anyone know why I cant use the second channel in a sip trunk, but I can use the first and third? |
| 19:47.04 | toresbe | sure, hang on |
| 19:47.37 | toresbe | http://pastebin.com/m78159757 <- here you go |
| 19:48.45 | toresbe | syzygyBSD: anything obviously wrong there? |
| 19:48.54 | syzygyBSD | toresbe: well, I noticed you didn't sanitize, and the secrets are different in your register line and the [chaos] ext |
| 19:49.11 | toresbe | I did sanitize, the passwords are made up |
| 19:49.21 | toresbe | I don't use passwords that weak IRL :) |
| 19:49.22 | syzygyBSD | ahh, ok |
| 19:49.37 | syzygyBSD | well, they are the same for both IRL right? |
| 19:51.29 | asagage | can anyone help me? the call goes through, but there is silence on both ends. |
| 19:52.49 | toresbe | syzygyBSD: yes |
| 19:52.55 | *** join/#asterisk BCS-Satori (n=somewher@75.148.21.113) |
| 19:53.55 | asagage | this is driving me nutz |
| 19:54.24 | BCS-Satori | Afternoon, this may be a tad off topic but I am looking for personal experience with carriers. Does anybody use a VoIP carrier that works well for them, that is not nexVortex or Vonage |
| 19:56.35 | syzygyBSD | toresbe: well, the issue is that [chaos] isn't registering with the server. I don't know why.... once you can get it to I think it will work (the register declaration does not say how to handle incoming calls) |
| 19:56.47 | toresbe | aha. |
| 19:59.20 | toresbe | So, this is an old/new syntax thing, or? |
| 19:59.54 | toresbe | register=> vs. [name]? |
| 20:00.49 | syzygyBSD | no, I have never used register=> because I never accept unauthenticated calls |
| 20:03.23 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 20:07.11 | Havokmon | Could someone look at my pastebin and tell me if I royally hosed something with my PRI? http://pastebin.com/m72f2246a |
| 20:07.31 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:07.53 | *** join/#asterisk nkoth_ (n=chatzill@bas4-ottawa23-1177650020.dsl.bell.ca) |
| 20:07.59 | toresbe | syzygyBSD: hmh |
| 20:09.21 | *** join/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 20:10.25 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:10.31 | *** part/#asterisk kratzers (n=kratzers@martha.pa.net) |
| 20:10.50 | De_Mon | BCS-Satori ~itsp |
| 20:10.55 | De_Mon | ~itsp |
| 20:10.56 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (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 |
| 20:11.41 | De_Mon | jbot tell BCS-Satori about itsplist-us |
| 20:12.03 | toresbe | I don't quite understand this. |
| 20:12.13 | syzygyBSD | Que? |
| 20:12.17 | toresbe | The register=> line is to update the registry at the opposite end to tell it where I am, right? |
| 20:12.29 | syzygyBSD | sure |
| 20:12.40 | syzygyBSD | but so should the friend declaration |
| 20:12.47 | toresbe | oh, it should? |
| 20:12.50 | syzygyBSD | yes |
| 20:12.50 | BCS-Satori | ~itsplist-us |
| 20:12.51 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com |
| 20:13.23 | nkoth_ | ~itsplist-ca |
| 20:13.23 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 20:15.10 | toresbe | syzygyBSD: it doesn't, though :) |
| 20:15.22 | toresbe | Are there any options for this in any other file than iax.conf? |
| 20:15.27 | syzygyBSD | indeed, you should find out why |
| 20:15.37 | *** join/#asterisk ddunavant (n=David@66.170.97.28) |
| 20:16.06 | syzygyBSD | ahh, |
| 20:16.21 | syzygyBSD | ok, there is no user= decleration for iax |
| 20:16.57 | iratik | When using attended call transfer, I press *2 and I get a prompt to enter the extension, how do I abort the transfer in case the extension is not there? |
| 20:16.59 | syzygyBSD | well, there is, but it should be the same as the [chaos] |
| 20:17.08 | syzygyBSD | look at http://www.voip-info.org/wiki/view/Asterisk+IAX+authentication |
| 20:17.08 | iratik | looked on http://www.voip-info.org/wiki/index.php?page=PBX+CallTransfer... found no answers there |
| 20:23.57 | *** join/#asterisk Porks (n=Porks@201.62.79.12) |
| 20:27.40 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 20:33.44 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:35.40 | toresbe | syzygyBSD: ok, it doesn't refuse calls now |
| 20:35.54 | syzygyBSD | yeah.... |
| 20:36.06 | toresbe | i defined the section as being the same as the user name and now things have changed. |
| 20:36.19 | syzygyBSD | dirty IAX |
| 20:36.20 | toresbe | I still have to use register=> for it to register, but it registers and it doesn't discard calls. |
| 20:36.36 | toresbe | Problem is, nor does it do anything else |
| 20:37.01 | toresbe | phones just disappear into nothing; and nothing happens on the asterisk command line even though I've invoked it with -vvvvvvv |
| 20:37.45 | anonymouz666 | Nobody is missing [TK]D-Fender? |
| 20:37.54 | syzygyBSD | who? |
| 20:38.00 | syzygyBSD | wow, he really is gone |
| 20:38.05 | syzygyBSD | unless it is you |
| 20:38.44 | anonymouz666 | no, he is not here. that's miracle. |
| 20:39.02 | syzygyBSD | maybe he took off this week, is vacationing somewhere tropical |
| 20:39.29 | *** join/#asterisk w3n0y (n=chatzill@241644hfc229.tampabay.res.rr.com) |
| 20:39.56 | anonymouz666 | somewhere like huntsville, alabama. to get even more near asterisk. |
| 20:39.57 | anonymouz666 | lol |
| 20:40.06 | *** join/#asterisk mmlj4 (n=jkelly@ip70-171-94-246.no.no.cox.net) |
| 20:44.20 | shido6 | hes over at #freeswitch |
| 20:44.24 | shido6 | :) |
| 20:44.57 | toresbe | It works outgoing now, anyway |
| 20:48.33 | *** part/#asterisk harryv (n=harry@0x55508034.adsl.cybercity.dk) |
| 20:50.13 | shido6 | oh gawd |
| 20:51.25 | toresbe | but not incoming |
| 20:51.27 | toresbe | >:( |
| 20:52.49 | *** join/#asterisk matsk (n=mk@83.233.97.210) |
| 20:56.24 | Kobaz | any idea why my polycom 320 would lock up if i dial something outside the digit map |
| 20:59.50 | *** join/#asterisk fugitivo (n=ajf@201-212-152-100.cab.prima.net.ar) |
| 21:00.00 | syzygyBSD | Kobaz: cuz I programmed it to? |
| 21:00.45 | lirakis | ~itsplist-us |
| 21:00.46 | jbot | [~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com |
| 21:00.51 | Kobaz | nope |
| 21:02.08 | ZaVoid | google presentation sux |
| 21:02.14 | ZaVoid | i just lost 30 mins of work |
| 21:02.22 | ZaVoid | and its not in my revisions |
| 21:02.32 | syzygyBSD | shoulda saved |
| 21:02.44 | ZaVoid | fender left? |
| 21:02.46 | ZaVoid | where he go? |
| 21:04.09 | MooingLemur | ~itsplist-ca |
| 21:04.10 | jbot | [~itsplist-ca] Here are some popular ITSPs (Canada) starting with the more respected ones : http://www.unlimitel.com , http://www.les.net , http://www.babytel.ca |
| 21:08.25 | hmmhesays | oh exchange I'm about ready to go nuts |
| 21:09.26 | De_Mon | hmm? |
| 21:09.29 | De_Mon | hmm hmm? |
| 21:09.33 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 21:11.48 | hmmhesays | De_Mon: I'm trying to get ms exchange configured, but everything I do I still get the same IVR from exchange |
| 21:18.57 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 21:22.52 | *** join/#asterisk Putzz (n=me@CPE001a707d4d4e-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 21:24.09 | russellb | hmmhesays: exchange with asterisk? |
| 21:29.42 | lirakis | later all |
| 21:30.02 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:31.34 | bkruse | russellb: res_microsoft? |
| 21:31.46 | russellb | bkruse: no ... |
| 21:31.55 | russellb | bkruse: but you can talk to exchange using SIP/TCP |
| 21:31.58 | russellb | which is in a branch |
| 21:32.07 | russellb | currently in testing, which is why i asked |
| 21:34.48 | *** join/#asterisk simonr (n=simonr@mail.ingleinsurance.com) |
| 21:34.53 | bkruse | russellb: interesting... |
| 21:35.28 | hmmhesays | russellb: yes |
| 21:35.52 | hmmhesays | I'm trying to configure something very simple and ms exchange is making it very difficult |
| 21:36.01 | hmmhesays | although the voice recognition stuff works great |
| 21:37.39 | davevg-btwtech | is anyone running res_jabber on trunk successfully? |
| 21:39.20 | hmmhesays | i use fs for my jabber needs |
| 21:40.08 | davevg-btwtech | was just wondering before i submit this bug, works on /branch/1.4 fine, but not in /trunk for some reason, wanted to make sure it wasn't something dumb i was doing |
| 21:43.38 | toresbe | Hrm. Say I dial into my PBX and end up in a context named 'fromc3'. |
| 21:44.43 | toresbe | Is it possible to receive DTMF codes and then forward to that extension? |
| 21:45.41 | *** join/#asterisk kkn088 (n=kikoun@77.204.209.7) |
| 21:47.39 | toresbe | That is, say I want to dial into my PBX, and strike a number, and have it forward me to another PBX with the supplied extension |
| 21:52.19 | davevg-btwtech | toresbe, see the applications Read, Background, and WaitExten |
| 21:52.48 | *** join/#asterisk fujin (n=aj@unaffiliated/fujin) |
| 21:53.22 | toresbe | davevg-btwtech: thanks! |
| 21:55.13 | Kobaz | anyone have any idea why my polycom 320 would lock up if i dial something outside the digit map |
| 21:56.16 | *** part/#asterisk ddunavant (n=David@66.170.97.28) |
| 21:57.13 | _x86_ | ok so what's it mean if a station gets voltage coming from the line, and the channel bank shows that channel off-hook when i take it off hook, but i hear no dialtone? |
| 21:57.40 | *** join/#asterisk truz_`24 (n=truz_`24@74-138-37-35.dhcp.insightbb.com) |
| 21:58.30 | *** join/#asterisk jengelh (n=jengelh@sovereign.computergmbh.de) |
| 21:59.14 | jengelh | <General complaint about Makefile strategy> |
| 22:02.16 | jengelh | http://jengelh.hopto.org/f/libiax2-includes.diff |
| 22:03.23 | *** join/#asterisk ThatKidKel (n=Kelvin@66.236.241.67.ptr.us.xo.net) |
| 22:03.42 | ThatKidKel | does anyone know where I can find a list of dialplan no-no's for 1.4? I'm about to bite the bullet and upgrade.. |
| 22:03.57 | russellb | just read UPGRADE.txt |
| 22:04.01 | ThatKidKel | k |
| 22:04.56 | jengelh | patch with -p3 |
| 22:05.27 | _x86_ | what would cause asterisk not to provide dialtone on whole group of FXS channels? |
| 22:05.32 | _x86_ | using fxo_ks signalling |
| 22:07.36 | _x86_ | anyone? |
| 22:07.54 | *** join/#asterisk simonr (n=simonr@mail.ingleinsurance.com) |
| 22:12.50 | _x86_ | gah |
| 22:12.50 | *** join/#asterisk obnauticus (n=obnautic@c-71-193-157-188.hsd1.wa.comcast.net) |
| 22:13.08 | _x86_ | channel bank auto-configured as "IMMD FXS" instead of "LOOP FXS" |
| 22:13.09 | obnauticus | What configuration file do I modify to configure Voicemail to email (SMTP, IMAP) settings? |
| 22:13.51 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 22:14.27 | _x86_ | [TK]D-Fender: hey man i need help :( |
| 22:14.39 | _x86_ | [TK]D-Fender: none of my stations can get dialtone from asterisk |
| 22:14.49 | _x86_ | [TK]D-Fender: there is voltage on the line, however |
| 22:15.23 | _x86_ | zapata.conf and zaptel.conf are both using fxo_ks and fxoks (respectively), and the channel bank is setup for LOOP signalling |
| 22:15.39 | _x86_ | [TK]D-Fender: channel bank can see when the stations go off-hook, but asterisk can not |
| 22:15.43 | _x86_ | [TK]D-Fender: any ideas? |
| 22:15.57 | [TK]D-Fender | _x86_, You already know what I want. |
| 22:16.10 | _x86_ | but it's not feasible :( |
| 22:16.33 | _x86_ | can you help me with what i have? |
| 22:17.17 | [TK]D-Fender | _x86_, PASTEBIN <------- |
| 22:17.19 | tzafrir_laptop | _x86_, do you see the channels listed in 'zap show channels'? |
| 22:17.28 | _x86_ | tzafrir_laptop: yessir |
| 22:17.36 | _x86_ | tzafrir_laptop: and zttool gives me OK on all interfaces |
| 22:17.48 | _x86_ | ztcfg -vvvv reports everything fine |
| 22:18.01 | tzafrir_laptop | when you pick up the phone, does asterisk think it is offhook? (zap show channel NNN) |
| 22:18.11 | [TK]D-Fender | _x86_, And you misread me. What I SUGGEST you use for equipment, VS what I expect if yuo want any advice about what you are using now are 2 completely seperate things |
| 22:18.17 | _x86_ | tzafrir_laptop: no |
| 22:18.37 | _x86_ | tzafrir_laptop: normally if i set verbose very high in CLI, it will show me when a station goes off-hook |
| 22:18.43 | _x86_ | tzafrir_laptop: it's not doing that |
| 22:18.54 | tzafrir_laptop | hmm, well I'm actually not ure it means anything |
| 22:19.39 | _x86_ | yeah but i have 7 other asterisk servers with a lot of analog stations, and they all do it |
| 22:19.52 | _x86_ | channel zap/blah went off-hook |
| 22:19.56 | _x86_ | or whatever... |
| 22:20.19 | tzafrir_laptop | next: go to lower layers |
| 22:20.32 | _x86_ | hmm... ok |
| 22:20.41 | _x86_ | such as? |
| 22:20.54 | tzafrir_laptop | hmm.. sorry, I'm not sure I have a good advise here |
| 22:21.04 | _x86_ | i would assume the cabling to be ok, since both the channel bank and zttool/ztcfg report OK on the T1 interface |
| 22:21.31 | _x86_ | asterisk is running as root, so i doubt it would be a permissions thing (i have seen that before though) |
| 22:21.45 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 22:24.36 | tzafrir_laptop | can you call the other way? |
| 22:24.52 | tzafrir_laptop | what happens if you use a different port? |
| 22:25.18 | *** join/#asterisk frenzy_ (i=user@unaffiliated/frenzy) |
| 22:25.30 | _x86_ | GOD DAMMIT |
| 22:25.35 | _x86_ | sangoma wont answer the god damn phone |
| 22:25.57 | _x86_ | keep getting allison's voice "the person at extension 110 is unavailable" |
| 22:26.01 | _x86_ | HOW FUCKING ANNOYINGH |
| 22:26.21 | rob0 | "Your call is important to us, just not important enough to actually hire enough people to answer it." |
| 22:26.52 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 22:26.53 | a1fa | omfg |
| 22:26.56 | a1fa | dont ever buy valcom |
| 22:27.31 | _x86_ | "your call is important to us, so much so, that we expect our support staff to NOT put their phones in DND or log out of the queue when they leave" |
| 22:28.33 | a1fa | omfg |
| 22:28.36 | a1fa | so fucking retarded |
| 22:28.45 | a1fa | all their specs are fake |
| 22:28.56 | a1fa | i bought a $400 speaker that only does multicast |
| 22:29.00 | a1fa | i can make one for $5 |
| 22:29.08 | a1fa | idiots |
| 22:29.17 | jengelh | well |
| 22:29.21 | *** join/#asterisk ReDNeQ (i=ReDNeQ@cpe-66-68-37-190.austin.res.rr.com) |
| 22:29.28 | jengelh | if you take it very precisely, have not you let yourself been fooled? :) |
| 22:30.24 | ReDNeQ | hello everyone happy holidays |
| 22:30.41 | a1fa | lol |
| 22:31.57 | ReDNeQ | well a1fa: a grumpy holidays to you! ;) |
| 22:32.33 | a1fa | dude |
| 22:32.40 | a1fa | i spent $400 on a shitty speaker that does multicast |
| 22:32.47 | a1fa | how retarded |
| 22:32.52 | a1fa | they advertised SIP speaker |
| 22:32.53 | a1fa | lol |
| 22:33.25 | jengelh | are you in the US? |
| 22:33.35 | a1fa | yes |
| 22:33.38 | jengelh | then sue |
| 22:33.46 | a1fa | i called the factory and support fucker didnt know clue |
| 22:34.01 | a1fa | i am about to go there and cut his fucking nuts |
| 22:34.22 | a1fa | this is ridicilous |
| 22:34.29 | a1fa | who the fuck uses multicast to communicate? |
| 22:34.55 | jengelh | noone, because it's not routed over the internet :p |
| 22:35.04 | *** join/#asterisk renkho (n=renkho@200.31.115.45) |
| 22:35.31 | a1fa | heh |
| 22:35.34 | bkruse | a1fa: rocks clusting |
| 22:35.38 | bkruse | clustering* |
| 22:35.41 | a1fa | some fucker said multicast is more efficient |
| 22:35.49 | a1fa | i dont care |
| 22:35.59 | a1fa | it doesnt work |
| 22:36.01 | a1fa | stupid fucks |
| 22:36.06 | bkruse | a1fa: Here is the problem with multicast, when described, it sounds OK, but when figuring out how it works, when it works, your like what?! |
| 22:36.27 | bkruse | so I understand where you are coming from, now you know who not to buy from, just try to return it |
| 22:37.08 | a1fa | bkruse : hehe.. i can see how multicast can be used in paging scenarios |
| 22:37.18 | a1fa | but besides cave-man paging system it sucks |
| 22:37.25 | bkruse | a1fa: sort of.... |
| 22:37.29 | bkruse | exactly, ha ha |
| 22:37.46 | bkruse | for rocks clustering, they use it to distribute the base of, instead of p2p, which kind of make since, its faster in their scenario |
| 22:37.53 | a1fa | i asked if their system supports tftp |
| 22:37.55 | bkruse | s/base of/base os/g |
| 22:38.00 | a1fa | guy asks what is that? |
| 22:38.16 | bkruse | are you serious? I am truely sorry man :/ |
| 22:38.27 | jengelh | right now so would someone pick up that libiax2 patch |
| 22:40.02 | a1fa | yeah m an |
| 22:40.06 | a1fa | never do business with valcom |
| 22:42.21 | a1fa | stupid fucks |
| 22:42.28 | a1fa | now they want to charge restocking fees |
| 22:42.35 | a1fa | i dare them |
| 22:42.37 | bkruse | a1fa: how much? |
| 22:42.37 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:42.39 | a1fa | $30 |
| 22:42.42 | bkruse | <20% i would just do it |
| 22:42.46 | a1fa | speaker was $380 |
| 22:42.54 | a1fa | but they falsley advertised their product |
| 22:42.56 | bkruse | count your losses and go somewhere else |
| 22:43.05 | a1fa | thats a bad business moto |
| 22:43.09 | bkruse | I agree |
| 22:43.11 | syzygyBSD | hmm, do their terms permit you from billing your hourly rate to identify the issue? |
| 22:43.13 | a1fa | i want every penny back invested |
| 22:43.14 | bkruse | you can report them if you would like |
| 22:43.26 | a1fa | bkruse : where should i report them> |
| 22:43.34 | a1fa | they didnt give me any shit on refund yet |
| 22:43.46 | a1fa | if i get any lip from them, i am going to open can of whoop ass |
| 22:43.51 | bkruse | a1fa: you can report them, gah whats the website |
| 22:44.06 | bkruse | its part of the better business beuro (sp) |
| 22:44.26 | syzygyBSD | consumerist.com? |
| 22:44.30 | bkruse | which is a great organization, imo, because they have 'these are bad business's website' |
| 22:44.42 | syzygyBSD | bureau |
| 22:45.16 | a1fa | thats why I asked for demo model before giving them my cash |
| 22:45.26 | syzygyBSD | I prefer reporting them to visa.... charging for services not provided. If enough people complain they can't do business anymore |
| 22:45.28 | a1fa | none of these son of bitches want to loan demo models |
| 22:45.29 | bkruse | syzygyBSD: thank you :] |
| 22:45.31 | syzygyBSD | BBB does nothing |
| 22:45.42 | bkruse | syzygyBSD: they do nothing, but it keeps others from doing the same. |
| 22:45.48 | a1fa | bkruse : i am calling my bank to stop the payment |
| 22:45.54 | a1fa | fuck that |
| 22:45.57 | bkruse | I always google and BBB someones website before buying, helps a lot |
| 22:46.06 | syzygyBSD | my chat client has spell check |
| 22:46.08 | bkruse | a1fa: you can try that.... |
| 22:46.11 | riddlebox | if you are in a conference room and person 2 conferences' person 3, is there a way to drop person 3? |
| 22:46.28 | bkruse | syzygyBSD: mine does too, but I disabled it when I was designing a pidgin plugin, have to re-enable it :] |
| 22:46.36 | syzygyBSD | I normally just stick with newegg, it has never let me down yet |
| 22:46.41 | bkruse | syzygyBSD: same |
| 22:47.03 | a1fa | ok i am pissed |
| 22:47.04 | a1fa | brb |
| 22:47.38 | syzygyBSD | heh, I have had way too much fun calling into support for other people |
| 22:48.27 | syzygyBSD | I know I am in no way associated with the purchase, so you can't tell me anything. Now that we have that settled, I will give you the info you need to fix it. Don't tell me anything, you arn't allowed to, just fix it |
| 22:49.20 | syzygyBSD | people get really confused, and it is easier to just do what I want then to not |
| 23:07.11 | *** join/#asterisk craigk (n=ckowald@58.174.150.119) |
| 23:30.17 | *** join/#asterisk RoyKa (n=roy@ip-236-9-149-91.dialup.ice.no) |
| 23:31.20 | *** join/#asterisk vetetix (n=vetetix@83.222.34.49) |
| 23:35.23 | LucanUK | right.. things have gone a lot better now.. I started all again with the old chan_mobile, now I am getting constant connect disconnet cycle with a bluetooth nokia (for use as a trunk) any known issues? |
| 23:47.36 | *** join/#asterisk Aces1up (n=dude@ip70-173-52-152.lv.lv.cox.net) |
| 23:48.24 | Aces1up | would u guys use skype for business purposes? thinking of using them with an 1-800 number... |
| 23:51.07 | fujin | no |
| 23:53.02 | *** join/#asterisk __freedom__lover (n=eduardo@201-43-197-219.dsl.telesp.net.br) |
| 23:53.26 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 23:53.31 | __freedom__lover | can someone tell me if i can create a exten with 'not' |
| 23:53.53 | __freedom__lover | ? |
| 23:54.15 | __freedom__lover | everything different of 444X, do this.. |
| 23:54.33 | fujin | what exactly do you mean? |
| 23:56.26 | *** join/#asterisk Notre1 (n=Notre1@c-76-123-120-21.hsd1.tn.comcast.net) |
| 23:56.57 | Notre1 | How do I change the "sender domain" in sendmail? |
| 23:57.35 | fujin | I think it's the hostname of the box, in sendmail |
| 23:57.37 | fujin | it's notoriously shit. |
| 23:57.39 | fujin | #sendmail |
| 23:58.11 | De_Mon | this isn't #sendmail! |
| 23:58.49 | Notre1 | Sorry, a sendmail newbie, and it is asterisk related. I'm trying to send out voicemail notifications. But, thanks for the info. |
| 23:58.52 | De_Mon | __freedom__lover I think exten => _XXXX would do that. |
| 23:59.04 | *** join/#asterisk zpertee (n=chatzill@123.sub-70-217-9.myvzw.com) |
| 23:59.15 | fujin | Notre1: install another not-shit MTA. |
| 23:59.18 | fujin | I'd suggest postfix |
| 23:59.24 | De_Mon | you may need to put that exteion either before or after _444X for it to work as desired |
| 23:59.27 | fujin | you can do a pretty basic satellite (send-only, no receive_ install. |
| 23:59.44 | Notre1 | fujin, thanks for the info. I'll give that a try |
| 23:59.51 | fujin | hell, dpkg even has a wizard for it in Ubuntu. |