| 00:02.38 | *** join/#asterisk tyler__ (n=tyler@ip70-190-145-81.ph.ph.cox.net) |
| 00:03.28 | MatBoy | does someone has a clue for FATAL: Module hfcpci not found. ? |
| 00:03.43 | MatBoy | I'm following docs, but this module is not loaded at all |
| 00:04.45 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 00:08.26 | jaytee | mmmm, I'm trying cardamom coffee for the first time and I'm liking it. |
| 00:09.30 | MatBoy | mhh, stupid modules every time :P |
| 00:13.07 | jaytee | chan_stupid: error opening channel. |
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| 01:04.11 | linagee | ManxPower: hahaha! i just thought of something. people pay money to have their numbers unlisted. what if i have no number to pay for to be unlisted? :-> |
| 01:04.21 | hsv-al | toeguhl |
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| 01:14.12 | TrentCreek | Ask jeeves |
| 01:17.15 | jeev | how cute |
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| 01:32.42 | TrentCreek | how Jeevy |
| 01:33.22 | jeev | cause i'm not jeeves |
| 01:34.30 | TrentCreek | oops |
| 01:34.36 | TrentCreek | Ask Jeev |
| 01:36.11 | hardwire | gets jeev's info.. |
| 01:36.15 | hardwire | jeev: right back at ya.. |
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| 01:37.43 | jeev | lol |
| 01:45.51 | *** join/#asterisk nickjqw (n=webbn@c-98-232-63-177.hsd1.wa.comcast.net) |
| 01:47.15 | nickjqw | Hello all, I'm trying to figure out why a new setup of asterisk can't make outgoing calls from a hardphone. Works fine from the console with 'dial 5555555' but not when dialing the same number from a handset. Incoming calls and handset to handset work fine. Any clue on where to look? |
| 01:48.06 | nickjqw | I get "Call from '01' to extension '5555555' rejected because extension not found" |
| 01:48.37 | nickjqw | Which is weird, since the "5555555" is matched on the dialplan from the console |
| 01:48.37 | jaytee | you don't have an outbound route in the context for that extension or that extension doesn't exist |
| 01:49.38 | nickjqw | how do I tell what context the hardphones are in? I added the outgoing route to "default". |
| 01:49.56 | jaytee | are the hardphones sip phones? |
| 01:50.07 | nickjqw | Yes hardphones are sip, outgoing route is is IAX |
| 01:50.33 | jaytee | what context does the hardphone you are using point to in sip.conf? |
| 01:51.23 | nickjqw | context=default (under "[general]" in sip.conf) |
| 01:52.02 | nickjqw | wait, I take that back. Under the specific extension, context=local-users |
| 01:52.25 | jaytee | does local-users have an outbound IAX route to dial? |
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| 01:54.42 | nickjqw | I'm not sure, learning as I go here. The outbound IAX route extensions are defined under "[outgoing]". |
| 01:54.48 | jaytee | as an example, in my convoluted dialplan I hacked together I have two contexts, one for local calls and one for long-distance calls. I include them in the contexts for other extensions. If a particular extension should be blocked from making long-distance calls it goes in a context that has not long-distance context included. |
| 01:55.21 | jaytee | try adding include=outbound in your local-users context |
| 01:55.23 | nickjqw | yep! |
| 01:55.27 | jaytee | then dialplan reload |
| 01:55.42 | nickjqw | That worked (I was lead to that by your suggestion before you typed it ) :) |
| 01:55.56 | jaytee | or include=outgoing. sorry :-) |
| 01:56.04 | jaytee | so it works now? |
| 01:56.10 | nickjqw | Yes, thanks jaytee! |
| 01:56.14 | jaytee | your welcome |
| 01:56.19 | jaytee | this is fun, isn't it? |
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| 01:57.04 | nickjqw | Actually, yes, just a bit frustrating. My old install was trixbox, trying to get away from that. This is my first install from scratch on a net5501... no more loud server :) |
| 01:57.26 | jaytee | one thing I learned that's important to remember is that a call will try to match against anything in a context and if no match is found then it will try any includes in the order they are listed. |
| 01:58.11 | beek | nickjqw: what distribution did you load on your net5501? |
| 01:58.27 | jaytee | I can't use trixbox as it restricts my dialplan logic and I have to do some tricky things to interface a Nortel Option 11c with * and have 4 digit dial between them so I've got kludges in the * dialplan and in the Nortel programming. |
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| 01:58.57 | ZarBla | I want to be able to achieve the following: I want to use my 2 standard lines at home. I want to plug them both in a Digium Card and be able to take any incoming calls from lets say line1 and redirect it to any number that I want using line2, but without loosing the call. I know I can achieve this but Im a little lost in FXO and FXS. I have a digium with 2 FXO. Can I achieve this? |
| 01:58.58 | nickjqw | jaytee: I head trixbox is near dead anyway, so decided to learn the real thing. It worked pretty well, though. |
| 01:59.27 | beek | ZarBla: yep... that's simple. |
| 01:59.53 | nickjqw | beek: I used ubuntu 8.04 server... nothing special. We use ubuntu on a lot of other servers at work, so it's what I'm familiar with. It has a laptop hard drive in it (SATA), so I didn't have to worry about any of the limiting ro issues of flash. |
| 02:00.04 | jaytee | ZarBla, yes you want 2 FXO ports |
| 02:00.17 | ZarBla | beek: Do you have any simple howto docs that explain how to do it? |
| 02:00.24 | jaytee | ~book |
| 02:00.25 | 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, or see ~buybook |
| 02:00.26 | nickjqw | beek: works great, now I can turn off my power sucking, noisy, heat creating old workstation become server thing. |
| 02:00.43 | jaytee | ZarBla, download the PDF |
| 02:00.50 | beek | nickjqw: I'm currently running * on an Asus WL-500gPremium but I'd like some more horsepower. |
| 02:01.11 | ZarBla | ok |
| 02:01.14 | beek | ZarBla: Just use the "Dial" function |
| 02:01.28 | nickjqw | beek: wow, that's impressive. I have one of those too, was experimenting with OpenWRT for QoS, etc., on it. I hated it for that. Now it just runs as an AP with DD-WRT... |
| 02:01.47 | beek | I have incoming calls to my home ring a phone in my home, a SIP phone on my desk at the office and my cell phone. Whichever picks up first wins, else it goes to voicemail. |
| 02:02.05 | nickjqw | beek: I've moved over to monowall on the PC Engines hardware instead of the WRT class boxes... it's $200 and a degree of magnitude less frustration. |
| 02:02.28 | beek | nickjqw: It works really well, but I'd like to add a database. |
| 02:03.01 | beek | nickjqw: I've gotten hooked on Vyatta for my router/firewall use. |
| 02:03.09 | nickjqw | beek: you might want to PC Engines hardware, I'm pretty impressed with it for the cost/benefit ratio. I don't know if they have SATA/IDE options, though. |
| 02:03.42 | nickjqw | beek: what do you load vyatta on or do you buy the pre-configured devices? |
| 02:04.12 | beek | nickjqw: I've purchased the 514s for small offices but in my office I have it loaded on an IBM 1u server. (I'd rather eat class than buy Dell) |
| 02:04.26 | beek | Vyatta preloads on Dell for their larger machine. |
| 02:04.40 | beek | Make that "eat glass" |
| 02:05.02 | nickjqw | beek: Have you used pfsense/monowall/ipcop? How does Vyatta compare? |
| 02:06.05 | beek | nickjqw: I've used the latter two. I like Vyatta because it's more "network geek" like. The configuration file is a single entitiy, so it's easy to edit with vi or save. |
| 02:06.21 | beek | There's a community edition with a pre-built VMware machine, if you'd like to look at it. |
| 02:06.39 | beek | e.g. You can get the installer CD to load on bare hardware or you can get the VM. |
| 02:07.04 | nickjqw | beek: I might try that out when I get a chance. It's showed up on the radar a few times, but never got the chance to try it. m0n0wall fits most my needs for now. If it had OpenVPN with LDAP authentication, I'd seriously consider it. |
| 02:07.43 | beek | You could load OpenVPN on it. It's a Debian derivative and you can load anything from the Debian repo. |
| 02:08.25 | wishes | hey anyone know much about using ngrep to monitor the commands comming/going from asterisk ? |
| 02:08.47 | nickjqw | That's promising. But sounds like I either need to buy their hardware or use a real server... We've got a lot of small offices, so the net5501s and PC Engines boxes are nice for that. |
| 02:09.33 | beek | nickjqw: Not really -- their 514 is built on a mini-itx board on a CF card. |
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| 02:10.39 | beek | nickjqw: I take it that this net5501 is an Intel-compatible processor? |
| 02:10.44 | trnzmeta | hi all |
| 02:11.02 | nickjqw | beek: Yes, AMD Geode 500MHz |
| 02:11.03 | trnzmeta | having difficulties when dialing disconnected telephone number |
| 02:11.19 | trnzmeta | the phone just dials and dials, doesn't recognised the disc signal |
| 02:11.26 | trnzmeta | anyone know what I should be looking at? |
| 02:12.15 | beek | nickjqw: Hmmm... I like their one-slot standard case. I could stick a TDM410 card in it. |
| 02:12.19 | Strom_C | trnzmeta: what does your Dial() line look like? |
| 02:13.05 | trnzmeta | off the conf or off the log file? |
| 02:13.14 | Strom_C | extensions.conf |
| 02:13.59 | nickjqw | beek: yes it's nice. I've had some quality control issues with the net5501, though, so beware |
| 02:14.31 | nickjqw | out of three boards one was bad, had it replaced 3 times, then finally got a board with only a minor issue I could fix myself... |
| 02:14.40 | beek | nickjqw: Oh... recently? |
| 02:14.50 | nickjqw | beek: yes, very recently |
| 02:15.57 | beek | No ventures to the pcengines site. |
| 02:16.26 | beek | nickjqw: you're happy with this little ALIX boards? |
| 02:16.36 | nickjqw | beek: Started months back, one board had a odd power on problem, wouldn't power up correctly after power loss. Took someone on site power cycling a few times to make it boot. Sent it back, same issue. Sent it back again, got new board, it had an issue with not being able to update the flash (or any CMOS setting). It worked though when not mounted in the case, so I was able to get it to work after re-mounting... so some kind of |
| 02:16.36 | nickjqw | short I suspect, but they should have caught that. |
| 02:17.49 | nickjqw | beek: so far, not much experience with them. Just bought one to throw monowall on for the home office. Works fine connecting up to a few IPSEC VPNs, etc., etc., just works. I'm not taxing it much though. I could have done it with a WRT like box, but I've had too much frustration with that in the past. |
| 02:18.19 | nickjqw | beek: Problem with the ALIX boards is no PCI slot, no SATA/IDE headers... but you could use asterisk on one with a CF. |
| 02:18.41 | beek | nickjqw: I'm looking at the site now. |
| 02:19.03 | Strom_C | trnzmeta: why is it taking you five minutes to paste a single line of text? |
| 02:19.03 | nickjqw | beek: also, make sure it will run linux unmodified... on the net5501 I can run anything a regular x86 can, but the ALIX board won't run the standard m0n0wall x86 image at all, needs the special ALIX build, not sure why. |
| 02:19.22 | nickjqw | beek: it should work, it's the same CPU as the net5501 and half the price. |
| 02:20.00 | beek | nickjqw: I see there is a 3.3V PCI slot (I thought PCI was part of the miniITX spec). I just haven't found the case that fits it. |
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| 02:21.36 | nickjqw | beek: IC, I didn't notice that on the board I have, but they may have other boardds |
| 02:22.00 | beek | nickjqw: Thanks for the info! That will give me some things to mull over. |
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| 02:23.02 | nickjqw | beek: no prob. I have the ALIX.2 board, the .1C has the PCI slot... |
| 02:24.29 | nickjqw | beek: later, have a good hunt |
| 02:24.40 | beek | nickjqw: GN |
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| 03:07.38 | Galaxor | Hi. I've got a basic sip routing question. I use a telephone to call my DID. Asterisk sees a sip call from my service provider. If I use Dial(SIP/my-soft-phone), it calls my softphone and I answer no problem. If I instead do Dial(otherphonenumber@serviceprovider-outbound), it rings the otherphonenumber, but when I answer it, I get no sound in either direction. |
| 03:08.07 | Galaxor | Am I doing something wrong, or is my service provider messed up? |
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| 03:22.26 | TJNII | ~sipnat |
| 03:22.27 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 03:22.35 | TJNII | Galaxor: ^^^^^ |
| 03:25.24 | Galaxor | I have a hard time believing it's nat problems. I can receive incoming calls to my softphone just fine. I can make outgoing calls from my softphone just fine. The only thing I can't do is bridge one phone to another. |
| 03:27.27 | TJNII | so two softphones behind your NAT work? |
| 03:29.10 | Galaxor | I haven't tried that. |
| 03:29.44 | Galaxor | What works is: Call from a landline, service provider sends it to my asterisk box (behind nat), who forwards it to softphone (behind nat). |
| 03:30.08 | Galaxor | Also working is: Softphone (behind nat) makes an outgoing sip call to service provider, who forwards it to landline. |
| 03:31.07 | Galaxor | What is not working is: Landline calls DID, service provider sends it to asterisk box (behind nat), who makes a sip call to service provider, who forwards it to landline. |
| 03:32.16 | mwalling | sounds like a clusterfsck ;) |
| 03:35.22 | Galaxor | The only difference is that, to dial my softphone, I say Dial(SIP/101), whereas to dial my landline, I say Dial(SIP/##########@vitel-outbound). It still rings the number I called. It just doesn't move any sound through. |
| 03:36.08 | TJNII | Sounds like a nat problem |
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| 03:39.59 | Galaxor | tjnii: Hm. Okay. I'll read those pages, then. Thanks. |
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| 03:41.18 | hmmhesays | anyone ever mess with nuera products? |
| 03:41.27 | hmmhesays | I have an old gateway here i"m looking for a manual for |
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| 04:07.41 | pputman | What would be a good solution to sip-proxy auth requests to a commercial sip switch? Openser? |
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| 04:16.46 | riddlebox | ~trixbox |
| 04:16.46 | jbot | [~trixbox] trixbox is a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org |
| 04:25.50 | riddlebox | ~thebook |
| 04:25.51 | 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 |
| 04:26.11 | riddlebox | sorry guys I am just answering some questions on the forums and need the info |
| 04:49.27 | ManxPower | riddlebox: you can /msg jbot and he will /msg you back |
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| 04:51.57 | ix33 | what's the best dialplan construct to use to test if a dialed extension contains a local exchange, given that i have a list of exchanges? |
| 04:52.55 | ManxPower | ix33: simple question, complex answer. Have you read The Book? |
| 04:53.42 | ix33 | ManxPower:yes, a while back. i just skimmed the two dialplan chapters and nothing jumped out. which topic should i specifically look at? |
| 04:53.53 | ix33 | (i mean, just now skimmed...) |
| 04:54.32 | ManxPower | There is nothing specific. you do it with dialplan pattern matching. If you understand that, you know how to do it. |
| 04:54.41 | ix33 | ok sure |
| 04:54.52 | ix33 | but i have a huge* list of extensions here |
| 04:55.05 | ManxPower | But expect it to be time consuming, complicated, and ultimately, useless. |
| 04:55.20 | ManxPower | ix33: exactly. |
| 04:55.42 | ix33 | ok so i've asked the wrong question. |
| 04:55.43 | ManxPower | do you have a large number of extensions or EXCHANGES? |
| 04:55.53 | ix33 | exchanges. |
| 04:56.21 | ManxPower | use the right words or things get too complicated |
| 04:57.11 | ManxPower | ix33: you can do pattern matching in the dialplan, you can put it all in a database, write a custom AGI script to return something to custom dialplan scripts, or you can get a provider with "all calls in area code are considered local and free" |
| 04:57.38 | ManxPower | Heck, the provider for my largest client doesn't even charge them for any calls within Louisiana or Mississippi |
| 04:58.12 | ManxPower | or you can go with a provider (usually VoIP) where ALL calls cost per min |
| 04:58.27 | ManxPower | a low cost, of course |
| 04:58.32 | ix33 | well unfortunately our PRI provider is set by contract already |
| 04:58.47 | ix33 | lesson learned |
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| 04:59.32 | riddlebox | ManxPower, I will do that next time sorry |
| 05:01.33 | ZarBla | I have a digium card with 2 fxo ports. I properly configured asterisk to take any call coming from FXO1 to call a number using FX02, using the dial command. Now as soon as it gets into the FX01 and starts to be forwarded, I have a big background noise and seems like I lost a lot of volume... Any idea how to fix this? I use a digium tdm402e... |
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| 05:24.15 | L8NteOwl | Hello.. anyone using Asterisk w/ MAX TNT for a Gateway ? |
| 05:24.19 | L8NteOwl | successfully ? |
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| 05:26.34 | d-tech | ne1 aware of a softphone that uses g.722 or any 16k solution that * supports |
| 05:29.02 | ManxPower | L8NteOwl: searching the mailing list archives, it's been talked about several times. |
| 05:29.08 | ManxPower | ~mailinglist |
| 05:29.09 | jbot | [~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search. |
| 05:31.20 | L8NteOwl | eyebeam supports g.722 if i'm not mistaken |
| 05:31.34 | L8NteOwl | http://www.counterpath.com/eyebeam.html |
| 05:31.51 | L8NteOwl | i have it installed but rarely use it |
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| 05:47.52 | d-tech | g.722 support in eyeBeam looks like a special OEM option ... there website notes it is not available in the Retail Version?! |
| 05:49.20 | d-tech | wow ... didn't think 16k support for a softphone was going to be this difficult?! |
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| 06:40.40 | Galaxor | <PROTECTED> |
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| 07:27.13 | *** join/#asterisk craigk (n=craigk@58.174.150.119) |
| 07:27.28 | *** join/#asterisk ludan (n=daniele@192.167.215.122) |
| 07:27.34 | ludan | hi guys |
| 07:28.53 | MCooper | I have a couple of questions concerning astericks and a project that I am working on? |
| 07:30.01 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 07:30.22 | MCooper | Could use some ideas... if there is anyone interested? |
| 07:30.52 | ludan | guys one of my users is «iptel/83644» and is the one that takes care of the connection with a toll free sip broker |
| 07:31.16 | ludan | sometimes the number is busy and to make it working I've to manually restart the whole server asterisk |
| 07:31.45 | ludan | is there a way to only disconnect and reconnect this user so that I can "restart" only the connections from landline? |
| 07:43.50 | tzafrir_laptop | notes the wrong spelling of MCooper |
| 07:44.11 | tzafrir_laptop | it's either Asterisk or asstricks |
| 07:44.38 | tzafrir_laptop | MCooper, anyway, ask your question |
| 07:45.56 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 07:46.06 | MCooper | I am working on a project to use asterisk as a blacklist maintainer. We are setting up a tip line and want to deal with the war dialers. |
| 07:46.53 | MCooper | The line is coming in from a CCME and being handed over to Asterisk to process, then either do a block and drop, or pass the call on to the tip center. |
| 07:47.16 | *** join/#asterisk alphanet (i=ircuser@shakotay.alphanet.ch) |
| 07:48.06 | alphanet | hello, is there any hope to run a B410P (4BRI) card with the Debian etch asterisk-bristuff (possibly with some slight changes to zaptel's qozap?). I don't want mISDN, and I would prefer to stick to Asterisk-bristuff 1.2 |
| 07:48.13 | MCooper | We have two E1s to handle the traffic. Dial tone is provided by the CCME. So we dial a number 999, verify the number making the call is not one that has called within the last 10 minutes... and if not forward to extension 130 |
| 07:49.25 | MCooper | The problem that we are running into is - we can dial the 999 and it dumps us back to a dial tone allowing us to dial the 130 extension.. we need that to be automated. |
| 07:49.42 | MCooper | tzafrir_laptop, That in a nutshell is the issue. |
| 07:54.51 | tzafrir_laptop | MCooper, can you pastebin the relevant parts of your dialplan? |
| 07:55.04 | tzafrir_laptop | the context into which it is dialed? |
| 07:56.35 | MCooper | This was a started project... |
| 07:56.56 | MCooper | after going over everything for two days, it just was not laid out the best. |
| 07:57.08 | MCooper | So I am in the process of starting it over from scratch... |
| 07:57.41 | MCooper | I could post what was there.. but I am thinking that if I build from scratch.. it would be a better design... |
| 07:58.17 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 07:58.58 | tzafrir_laptop | MCooper, you want to get a dialtone after dialing 999 or don't want to? |
| 08:01.38 | *** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk) |
| 08:01.52 | MCooper | No dial tone - just goto extension 130 |
| 08:05.22 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:09.05 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 08:09.05 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 08:09.37 | bkruse | needs to go to sleep |
| 08:10.48 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 08:11.41 | mvanbaak | bkruse: no |
| 08:12.06 | mvanbaak | not acceptable |
| 08:12.13 | bkruse | mvanbaak: ! |
| 08:12.15 | bkruse | How was the trip? |
| 08:12.24 | mvanbaak | it was GREAT |
| 08:12.35 | bkruse | mvanbaak: I am getting owned on some mxml stuff, I need to revert back and see how I originally parsed the tree. Something we added screwed things up |
| 08:12.37 | mvanbaak | lots of wind, not too much sun |
| 08:12.44 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 08:12.55 | mvanbaak | bkruse: yeah, I saw that in your last commit |
| 08:13.06 | bkruse | fun trip though? |
| 08:13.20 | mvanbaak | yeah |
| 08:13.29 | bkruse | good good |
| 08:13.36 | mvanbaak | wonderful sailing weather |
| 08:15.04 | mvanbaak | snuff-home had a good suggestion. We need to document the return value of functions etc |
| 08:15.11 | mvanbaak | I added it to the bug he created |
| 08:15.18 | mvanbaak | he did some work on app_amd and app_fax |
| 08:16.30 | mvanbaak | hhmm, the parsing here is working |
| 08:16.45 | mvanbaak | but I have a revision from before you started messing with it ;) |
| 08:17.24 | mvanbaak | but like russell said, we really need an ok on the schema first |
| 08:22.15 | *** join/#asterisk qdk (n=qdk@85.235.253.139) |
| 08:22.29 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 08:25.48 | bkruse | reads |
| 08:26.35 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 08:26.48 | mvanbaak | I should add that <return> node to the xmldoc.txt |
| 08:29.30 | *** join/#asterisk ^shark_ (n=Administ@41.222.2.65) |
| 08:29.44 | ^shark_ | hi friends,just a simple question,i am quoting for the 7960G IP phone for a sip voip solution,how important is it to quote for the licenses too |
| 08:30.28 | mvanbaak | you need the license. otherwise you are not allowed to use the phone (legally) |
| 08:34.13 | Uatec | how important? is that like "how important is it to pay for dvds in the shops?" |
| 08:39.46 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 08:40.02 | tzafrir_laptop | Uatec, you mean the $1 for the blank one? sure you pay for it ;-) |
| 08:40.29 | *** join/#asterisk ToTo (n=ToTo@207.176.6.190) |
| 08:41.16 | ^shark_ | mvanbaak: i suppose the ip phone comes with sip firmware? |
| 08:41.17 | Uatec | well if you're going to be reselling this system, you might like to be licenced for it... |
| 08:42.47 | ^shark_ | mvanbaak: .. and the sip firmware comes attached to the IP phone package, is it not? |
| 08:44.11 | *** join/#asterisk hi365 (n=hi365@bzq-219-141-66.static.bezeqint.net) |
| 08:44.16 | hi365 | ~bot |
| 08:44.16 | jbot | I ain't no stinkin' bot. I am a finely tuned and hand crafted tool. Oh wait... I guess I am a bot (that you should not abuse). |
| 08:44.24 | *** join/#asterisk implicit (n=bayan@ip72-211-213-26.oc.oc.cox.net) |
| 09:01.13 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 09:01.34 | puzzled | hi |
| 09:03.17 | bkruse | mvanbaak: there? |
| 09:03.30 | bkruse | I am going to revert, see how it is working |
| 09:03.59 | bkruse | it all b0rked when I did that conversion from infunction to an mxml_find_documentation("application", "Dial", "description"); function |
| 09:04.21 | bkruse | mvanbaak: Drop me an email and tell me what revision, gotta go to sleep |
| 09:16.28 | *** join/#asterisk Rico29 (n=Rico@sto93-4-88-162-214-225.fbx.proxad.net) |
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| 09:28.09 | *** join/#asterisk JPASS (n=james@host-87-74-7-50.dslgb.com) |
| 09:28.16 | JPASS | Hi All, |
| 09:28.21 | JPASS | Any guru's out there? |
| 09:28.56 | JPASS | I have a server that keeps failing - the line goes down. It's a server issue as a reboot sorts it out. Funnily a restart of ztcfg and asterisk doesn't/ |
| 09:32.16 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
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| 09:35.41 | *** join/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 09:36.33 | *** join/#asterisk mandh (n=mandh@82.137.216.38) |
| 09:38.57 | mandh | Hi all , can i put wav files as music on hold ? |
| 09:39.33 | tzafrir_laptop | JPASS, what hardware is it, exactly? What versions of Asterisk and Zaptel? |
| 09:39.51 | *** join/#asterisk cabbiepete (n=cabbiepe@83.244.133.169) |
| 09:42.09 | JPASS | Asterisk BE |
| 09:42.14 | JPASS | Laterst zaptel |
| 09:42.16 | Uatec | eurgh, BE? |
| 09:42.21 | Uatec | kills self |
| 09:42.28 | JPASS | huh |
| 09:42.28 | Uatec | is an Asterisk BE user |
| 09:42.40 | JPASS | What? What's wrong with BE |
| 09:43.38 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-9c6ddf941f1443f4) |
| 09:43.39 | JPASS | and a TE122B |
| 09:43.42 | Uatec | rpath is crap, BE is version 1.2 or something... |
| 09:43.51 | JPASS | Well we use RHEL5 |
| 09:44.09 | JPASS | BE is 1.4 on the C branch. Please don't contribute if you don't know what you're on about |
| 09:44.26 | *** part/#asterisk rcy (n=rcy@S010600003981572c.vc.shawcable.net) |
| 09:44.43 | Uatec | fuck you JPASS, fuck you |
| 09:44.50 | JPASS | Loser! |
| 09:44.57 | Uatec | wondesr if he's developing some kind of irc tourettes then realises that he's just being a dick |
| 09:45.02 | Uatec | sorry |
| 09:45.22 | *** join/#asterisk khronos (n=khronos@aquaman.perryinstitute.org) |
| 09:45.22 | Uatec | the version of BE that i've been using is 1.2 |
| 09:45.32 | JPASS | Yes that BE 'B@ |
| 09:45.34 | JPASS | B |
| 09:45.39 | Uatec | yes, tis |
| 09:45.44 | JPASS | but C is out which is on the 1.4 branch |
| 09:45.59 | tzafrir_laptop | Uatec, do they have a perl branch? |
| 09:46.01 | Uatec | i can't update it becuase like two of my systems can't route toconary |
| 09:46.03 | JPASS | Also you don't have to use rpath, they RPMs are all there for installation on a decent os. |
| 09:46.10 | Uatec | yeah, true |
| 09:46.28 | Uatec | my boss got ABE and said "run it on the OS it comes with, that makes better business sense" |
| 09:46.35 | JPASS | Anyway - why am i talking to you - you just told me to fuck myself! |
| 09:46.43 | Uatec | lol, i did, sorry about that |
| 09:47.17 | Uatec | now i use ubuntu and 1.4 |
| 09:47.19 | Uatec | is it 1.4? |
| 09:47.22 | Uatec | checks |
| 09:47.41 | JPASS | tzafrir_laptop - can you assist with my issue? |
| 09:48.13 | JPASS | I think the reboot made a difference over a restart of the processes as a reboot resets the PCI bus. |
| 09:48.29 | JPASS | This would point to a hardware issue - BUT... the issue only happens in the middle of the night! |
| 09:48.38 | JPASS | WOrks fine when it's being used |
| 09:48.50 | JPASS | during the dat |
| 09:48.54 | JPASS | day |
| 09:48.56 | JPASS | argh! |
| 09:48.56 | tzafrir_laptop | What do you mean by "line going down"? |
| 09:49.07 | JPASS | Well 'zap show status' shows OK |
| 09:49.20 | JPASS | but a PRI SHOW SPANS shows as down |
| 09:49.23 | tzafrir_laptop | What problem do you see? |
| 09:49.30 | JPASS | all we get on the PRI DEBUG SPAN 1 is.... |
| 09:49.43 | JPASS | Sending Set Asynchronous Balanced Mode Extended |
| 09:49.47 | JPASS | over and over. |
| 09:49.54 | tzafrir_laptop | Maybe it's a problem with the line, or with the provider or something? |
| 09:50.04 | JPASS | but then why does a reboot fix it? |
| 09:50.09 | JPASS | i can;'t explain that |
| 09:50.13 | tzafrir_laptop | Do you have any alarm on that line? |
| 09:50.25 | JPASS | no, it says OK |
| 09:50.29 | JPASS | zttool - OK |
| 09:51.29 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:56.31 | *** join/#asterisk bootc (n=bootc@78-33-44-114.no-dns-yet.enta.net) |
| 09:56.53 | bootc | hey folks, I have some Snom 320 phones and I'm trying to do an auto-answer on an Originate |
| 09:58.45 | bootc | what I do is I originate to Local/exten@context, then send to exten@context |
| 09:59.36 | bootc | in the first context I SIPAddHeader("Call-Info: sip:phones.adl;answer-after=0"), and on the phone I have auto_connect_type=headset |
| 10:00.05 | bootc | but what happens is the headset picks up, rings for a second or two, then the remainder of the call happens on speakerphone (unless you press the headset button) |
| 10:00.41 | tzafrir_laptop | JPASS, maybe something related to resetinterval? |
| 10:01.05 | tzafrir_laptop | (In zapata.conf) |
| 10:02.13 | JPASS | let me check.... |
| 10:03.20 | JPASS | resetinterval=never |
| 10:06.41 | tzafrir_laptop | new look for the wiki |
| 10:06.47 | tzafrir_laptop | http://voip-info.org/ |
| 10:06.56 | tzafrir_laptop | Does the search work for anybody? |
| 10:09.02 | *** part/#asterisk heh_v_water (n=heh_v_wa@mail.mtfreetech.us) |
| 10:10.05 | creativx | search works |
| 10:12.55 | *** join/#asterisk gfather1 (n=enforcer@79.173.240.157) |
| 10:17.32 | tzafrir_laptop | maybe it requires javascript? |
| 10:18.25 | *** join/#asterisk dna (n=dna@85.93.10.9) |
| 10:19.20 | MCooper | Uatec, did you have any issues with the Ubuntu install? |
| 10:19.29 | Uatec | nope. Why, are you? |
| 10:19.57 | JPASS | Hardy heron has issues with misdn comp;iliation. |
| 10:20.26 | Uatec | my ubuntu installation was sip only |
| 10:20.29 | Strom_C | JPASS: congratulations. are you going to file a bug or just complain? |
| 10:20.35 | JPASS | Issue with the way a variable is defined as it's already defined in the kernel... |
| 10:20.42 | MCooper | Yeah.. Coming up with termcap issues during the ./configure process... |
| 10:20.48 | JPASS | I'm not complaining, just helping. I use RHEL |
| 10:21.10 | JPASS | We didn't find a solution unfortunately |
| 10:21.18 | MCooper | Looked everything over... pissing me off.. |
| 10:21.20 | JPASS | Did you consider Centos |
| 10:21.31 | MCooper | Yeah... |
| 10:21.56 | JPASS | (PS Usefull comments Strom_C - really helpful!) |
| 10:22.14 | JPASS | Sorry i can't help more MCooper - |
| 10:22.19 | JPASS | use centos is all i can add!! |
| 10:22.25 | puzzled | JPASS: there's an misdn mailing list. do you report the problem there? |
| 10:22.50 | puzzled | JPASS: http://lists.beronet.com/cgi-bin/mailman/listinfo/misdn-asterisk |
| 10:23.17 | JPASS | no - we were'nt fussed on the OS too much, just used centos instead. Lazy but easy! |
| 10:23.27 | MCooper | Yeah... |
| 10:23.35 | MCooper | I guess I will go grab it... |
| 10:24.27 | MCooper | later everyone... |
| 10:25.20 | thomas | if I would like uninstall asterisk |
| 10:25.38 | thomas | which directory I should remove? |
| 10:25.49 | thomas | <PROTECTED> |
| 10:26.13 | khronos | /var/spool/asterisk /etc/asterisk |
| 10:26.25 | thomas | khronos: and the binarys? |
| 10:26.33 | khronos | the safe_asterisk or init.d script if you installed these. |
| 10:26.51 | khronos | and /var/lib/asterisk |
| 10:27.16 | thomas | is asterisk-addons-1.4.7 the newst? |
| 10:27.25 | thomas | latest.. |
| 10:27.28 | khronos | yes, I beleive so for the 1.4 branch. |
| 10:28.05 | thomas | khronos: how i can only compile cdr mysql? |
| 10:28.09 | thomas | i need only this. |
| 10:28.32 | khronos | Actually I'm in a similar situation myself. |
| 10:28.42 | khronos | I'm looking for input on which logging module I should use. |
| 10:29.00 | khronos | what's the difference in using the odbc module to log to a mysql database or the mysql module from the addons package? |
| 10:29.14 | thomas | i dont know.. hm. |
| 10:29.34 | khronos | also if I eventually want to use the real-time way of doing things what module would I want to use? |
| 10:30.03 | khronos | Should I use odbc all the way around or mysql module from addons for cdr logging? |
| 10:30.26 | thomas | hmm. |
| 10:30.31 | thomas | i use mysql. :) |
| 10:30.46 | khronos | if you only want the mysql module from addons I think you may be able to slect this in make menuconfig from with in the addons package, but I haven't gotten that far on this system I'm working on just yet. |
| 10:33.17 | jblack | khronos: odbc is where it's going. |
| 10:33.41 | jblack | used to be each database got it's own resources. Now, asterisk just uses one, odbc, and relies on the odbc drivers. |
| 10:33.52 | thomas | ah, yes. make menuconfig |
| 10:33.54 | jblack | though it still supports the older ones. |
| 10:34.14 | hi365 | ~centos52bug |
| 10:34.15 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 10:35.58 | *** join/#asterisk Strom_M (n=strom@208.127.172.112) |
| 10:38.18 | khronos | Ok, so I'm better off using odbc for my logging and realtime? |
| 10:40.30 | jblack | yeah. |
| 10:41.06 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
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| 10:50.56 | khronos | K. |
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| 10:58.50 | *** join/#asterisk stimpie (n=stimpie@84-104-6-10.cable.quicknet.nl) |
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| 11:03.30 | gfather1 | guys could u give me some guidelines |
| 11:03.48 | gfather1 | like i have 7 analoge phones to the company i work for |
| 11:04.01 | gfather1 | how should i connect them to the Asterisk server ? |
| 11:05.03 | alphanet | I would connect them through 4 ATA such as the Linksys-dual port at a total cost of about 400 EUR. Alternative: replace them with SIP phones if it's practical (wireful), or even a channel bank |
| 11:06.46 | gfather1 | what i mean is |
| 11:06.57 | gfather1 | that these line are from the phone company |
| 11:06.57 | MatBoy | mhh ok, I have elastix installed, my ISDN card is seen |
| 11:07.08 | MatBoy | now itÅ in TE mode and I need to set it in NT mode |
| 11:07.13 | gfather1 | and i want to connect them to the Asterisk server |
| 11:07.27 | alphanet | ah ok, then you could either use FXO ATAs, or digium analog cards, or channel banks. I am more used to ISDN myself |
| 11:07.32 | gfather1 | and make the inner connections of ip phones |
| 11:07.46 | gfather1 | <alphanet> oks cool |
| 11:07.47 | gfather1 | brb |
| 11:07.55 | alphanet | I don't have much experience with analog outside lines |
| 11:09.31 | alphanet | gfather1: http://www.voip-info.org/wiki/view/Asterisk+Channel+Bank |
| 11:10.32 | *** join/#asterisk Datax (n=john@88.191.19.100) |
| 11:11.25 | Datax | Hi all, I'm looking in to what Asterisk can and can't do from a STUN point of view |
| 11:11.41 | Datax | what exactly is possible ? |
| 11:12.29 | *** join/#asterisk CVirus (n=GoD@196.218.41.31) |
| 11:14.09 | MatBoy | what do I have to do when I have a down link ? Port 1 Type NT Prot. PMP L2Link DOWN L1Link:DOWN Blocked:0 Debug:0 |
| 11:16.43 | CVirus | Why can't my system see my second FXO X100P card ... ZT_CHANCONFIG failed on channel 2: No such device or address (6) |
| 11:19.12 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 11:19.58 | *** join/#asterisk Segnale007 (n=Segnale0@host115-10-dynamic.33-79-r.retail.telecomitalia.it) |
| 11:24.12 | MatBoy | when I have a ISDN card in NT mode and connected to an ISDN pbx, should that stack be UP automaticly ? |
| 11:27.47 | puzzled | MatBoy: you need to have the appropriate drivers installed, configured and configure stuff in asterisk etc. |
| 11:30.34 | *** join/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 11:31.34 | *** join/#asterisk ibnolqaiyem (n=ibnolqai@41.196.251.156) |
| 11:31.46 | JT | MatBoy: using what drivers...? |
| 11:31.55 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:32.01 | ibnolqaiyem | what is good documentation for learn asterisk? |
| 11:32.23 | MatBoy | puzzled: JT uhm, on a elastix install I only started misdn |
| 11:32.34 | MatBoy | and the card is recognized well |
| 11:32.52 | puzzled | MatBoy: sorry I'm not familiar with elastix |
| 11:33.37 | caio1982 | ibnolqaiyem: http://tfot.leifmadsen.com also known as 'the book' |
| 11:33.40 | MatBoy | puzzled: no problem :) |
| 11:35.19 | puzzled | MatBoy: you may need to add power to the NT line with an NT-1 or something else that can provide power to an ISDN line |
| 11:35.53 | ibnolqaiyem | are there some hardware basics i need to learn asterisk? |
| 11:35.55 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 11:36.37 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 11:36.37 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 11:36.52 | MatBoy | puzzled: mhh, but I just crossconnected the NT card to the isdn PBX |
| 11:36.57 | MatBoy | should be OK I thought |
| 11:37.13 | puzzled | iirc that depends on the amount of power it needs to provide |
| 11:38.01 | MatBoy | puzzled: mhh, but an NT1 box just between that cable you think ? |
| 11:38.25 | puzzled | MatBoy: if you have one you can easily try it |
| 11:38.49 | JT | pbxes do not require power from the line |
| 11:39.02 | MatBoy | puzzled: yes I have one, but should I use rj45 for both do you think ot the rj11 (line) for the isdn card ? |
| 11:39.12 | MatBoy | JT: no, but the card shows down |
| 11:39.27 | JT | MatBoy: model of card? |
| 11:39.45 | MatBoy | JT: Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) |
| 11:39.49 | MatBoy | simple one |
| 11:40.02 | JT | you will need a BRI crossover cable |
| 11:40.19 | MatBoy | JT: normal lan crossover ? |
| 11:40.22 | JT | no. |
| 11:40.24 | JT | you will need a BRI crossover cable |
| 11:40.27 | MatBoy | ok :) |
| 11:40.29 | MatBoy | let me make one |
| 11:40.38 | MatBoy | actually |
| 11:40.41 | MatBoy | I think I have one |
| 11:40.53 | alphanet | MatBoy: the last time I configured an HFC-s card with Asterisk in NT mode: 1. I was using zaptel/zaphfc, not mISDN; 2. it went UP automatically, without being connected to the other party. 3. I used a ISDN cross-over cable. 4. I only needed power when using ISDN phones. |
| 11:41.14 | JT | misdn sucks btw |
| 11:41.41 | alphanet | well, the only advantage of mISDN in my opinion is that it seems better than zaptel + app_fax |
| 11:42.02 | alphanet | however, for pure voice operation, it sucks: 2.6 kernel only, will be replaced, etc. |
| 11:42.09 | MatBoy | JT: nah, with elastix it seems to be quite ok |
| 11:42.17 | alphanet | I use zaptel since 2005 with a lot of success |
| 11:42.50 | JT | MatBoy: i'm telling you the reality, the elastic pretties don't make a driver any better |
| 11:42.53 | JT | misdn sucks. |
| 11:42.57 | *** join/#asterisk ming_zym (n=ming_zym@222.128.39.136) |
| 11:46.25 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 11:48.38 | MatBoy | JT: yes normally it does, but the way it enables was in elastix much better than any other way |
| 11:48.48 | JT | uhuh |
| 11:48.55 | JT | it's especially bad with NT mode |
| 11:48.57 | MatBoy | can't help it :) |
| 11:49.01 | *** join/#asterisk Superbartt (n=bart@ipd50a21c9.speed.planet.nl) |
| 11:49.17 | JT | there are other options. |
| 11:49.34 | MatBoy | JT: like ? |
| 11:49.56 | Superbartt | hmmfg, _X. vs _[6-8]X in the same context, _[6-8]X would get a higher priority ("first choise") right? |
| 11:50.17 | JT | MatBoy: zaphfc with zaptel of zaphfc with bristuff |
| 11:51.24 | MatBoy | JT: I have the idea that I can use and NT1 box for the crossover because this has termination already |
| 11:51.56 | JT | well that's wrong |
| 11:52.11 | *** join/#asterisk eth01 (i=foo@gentoo/user/eth01) |
| 11:52.38 | *** part/#asterisk dominic1 (n=dob@213.221.82.242) |
| 11:54.00 | *** join/#asterisk PepOSX (n=angeldav@190.72.146.157) |
| 11:55.01 | MatBoy | JT: I need to resistors for sure ? |
| 11:55.22 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 11:55.23 | *** mode/#asterisk [+o russellb] by ChanServ |
| 11:56.04 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 11:58.31 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 11:59.27 | *** join/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 12:06.57 | *** join/#asterisk ToTo (n=ToTo@209.8.41.34) |
| 12:09.19 | MatBoy | JT: works ! thanks ! |
| 12:09.24 | MatBoy | that crosscable |
| 12:09.32 | *** join/#asterisk ArashHemmat (n=arash@91.184.89.19) |
| 12:10.43 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:12.57 | *** join/#asterisk beek (n=klinebl@65.211.106.242) |
| 12:13.23 | MatBoy | ok, now I need to be able that my BRI uses my sip account that I have setup as trunk to dail out |
| 12:17.26 | [TK]D-Fender | ~book |
| 12:17.27 | 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, or see ~buybook |
| 12:17.31 | [TK]D-Fender | MatBoy: ^^^^ |
| 12:17.45 | MatBoy | [TK]D-Fender: yep thanks ! |
| 12:17.57 | [TK]D-Fender | MatBoy: you need to understand the dialplan. This is the most important part of * |
| 12:18.13 | MatBoy | [TK]D-Fender: yes I know about that |
| 12:18.25 | MatBoy | but this "feels" different in some strange way |
| 12:19.24 | MatBoy | [TK]D-Fender: actually I would assume that elastix did had some more detailed info about their panel |
| 12:19.59 | *** join/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 12:20.30 | [TK]D-Fender | MatBoy: Oh.... another GUI user. #freepbx is waiting for you. Move along now... |
| 12:21.18 | MatBoy | [TK]D-Fender: yes I can work this out :) |
| 12:21.19 | MatBoy | thanks ! |
| 12:21.33 | bkruse | [TK]D-Fender: :) |
| 12:21.53 | [TK]D-Fender | bkruse: mornin' |
| 12:22.23 | bkruse | [TK]D-Fender: Good morning. GUI users giving you much trouble? |
| 12:22.48 | russellb | bkruse: GUI users offend him at a deep personal level :-p |
| 12:22.49 | [TK]D-Fender | bkruse: Not that much lately. |
| 12:22.59 | bkruse | good good good |
| 12:23.11 | bkruse | russellb: I understand that, and I try to grab as many as i can away :D |
| 12:23.51 | [TK]D-Fender | russellb: No. Not GUI users in general. Just the ones that feel we are here to discect whatever junk it generated and reverse engineer it to learn what 3 blanks in a stupid for a monkey should be able to fill out :) |
| 12:24.10 | bkruse | [TK]D-Fender: Which is a large percentage :P |
| 12:24.41 | [TK]D-Fender | bkruse: So... its just the moronic ones that don't get the picture that this is not 2nd level GUI support. |
| 12:24.52 | [TK]D-Fender | bkruse: Make your bed, and lie in it. |
| 12:24.58 | [TK]D-Fender | lay* |
| 12:25.08 | [TK]D-Fender | Darn homonym mixup :) |
| 12:25.36 | bkruse | [TK]D-Fender: Exactly. Those, deserve the wrath |
| 12:26.16 | [TK]D-Fender | bkruse: And to think I've only kicked ONE person out of here, and that's for spamming the channel :) |
| 12:26.48 | alphanet | what is a GUI? |
| 12:27.12 | [TK]D-Fender | bkruse: I'm thinking of starting an #asterisk IRC User Shit-list to ID those who try to leech support while not even USING *, or lurk around for support while using GUI's :) |
| 12:27.15 | [TK]D-Fender | ~gui |
| 12:27.16 | jbot | it has been said that gui is (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, ... or at http://pla-netx.com/linebackn/guis/index.html. Of course Real Programmers use the command line interface. See cli |
| 12:27.37 | alphanet | :) |
| 12:27.55 | [TK]D-Fender | alphanet: In the case of context, and interface that builds your * configs for you folling its own cookie-cutter logic and building undecipherable garbe |
| 12:28.06 | [TK]D-Fender | garbage* |
| 12:28.14 | alphanet | ah ok, I have done that (tm) |
| 12:28.21 | *** join/#asterisk s0lid (n=s0lid@122.53.110.157) |
| 12:28.26 | alphanet | TWiki -> PostgreSQL -> config generator |
| 12:28.28 | alphanet | :-> |
| 12:28.31 | [TK]D-Fender | alphanet: Built, or used? |
| 12:28.36 | alphanet | built |
| 12:28.39 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 12:28.44 | [TK]D-Fender | alphanet: For *? |
| 12:28.45 | *** join/#asterisk DarnoQ (n=d@chello089076192243.chello.pl) |
| 12:28.50 | alphanet | yes |
| 12:29.01 | [TK]D-Fender | alphanet: Using a WIKI front end? Sounds interesting. |
| 12:29.03 | alphanet | it allows people to use TWiki to configure some of asterisk |
| 12:29.17 | alphanet | yes, it's a bit weird |
| 12:29.46 | bkruse | [TK]D-Fender: Where is that button in the GUI!? |
| 12:30.26 | [TK]D-Fender | bkruse: You meant he "Go" button that they all assume is out there? :) |
| 12:30.29 | alphanet | but that company has a "everything in TWiki" policy |
| 12:30.38 | DarnoQ | could someone please advice me on how to record g729 call into wav or mp3 ? when i try to record using monitor i get an error " file.c:199 ast_writestream: Unable to translate to format pcm, source format g729". I got 2 licences for testing but the call is in passthrough |
| 12:30.39 | alphanet | which is in its way logical |
| 12:31.14 | alphanet | ok, I need to go, thanks for support :) |
| 12:31.18 | alphanet | have a nice afternoon |
| 12:31.26 | [TK]D-Fender | alphanet: Every time I here the term "corporate policy", the statistical evaulation of its practical implication is (not pulling punches) "fucking stupid" |
| 12:31.52 | bkruse | [TK]D-Fender: I want to do this convoluted asterisk setup with crazy call matching patterns, misdn, t1, through 2 channels banks and a cisco gateway, where is the gui page for that? |
| 12:31.52 | alphanet | well, no, in this case it's quite logical |
| 12:31.57 | alphanet | they mainly evolve around documentation |
| 12:32.06 | alphanet | so everything is documentation for them, including asterisk |
| 12:32.13 | alphanet | and you get a version control for free |
| 12:32.23 | alphanet | that the user can manage if needed |
| 12:32.37 | alphanet | see it as a complex way to store your config data in CVS |
| 12:32.46 | alphanet | :) |
| 12:32.59 | [TK]D-Fender | alphanet: Documentation is GREAT. Thinking you can use a teaspoon to fill the role of a socket-wrench - is NOT :p |
| 12:33.05 | alphanet | :->> |
| 12:33.21 | alphanet | well, I use asterisk to open the company's door |
| 12:33.24 | [TK]D-Fender | alphanet: Yeah I can see how you might parse out a WIKI for it... it IS a nifty idea actually.. |
| 12:33.45 | [TK]D-Fender | alphanet: Do you do raw dialplan in the and WGET+parse it into a primary template? |
| 12:34.04 | alphanet | no, I use the Wiki format (the Wiki language itself) |
| 12:34.05 | [TK]D-Fender | there* |
| 12:34.17 | macros73 | Hey now, I'm a stupid for a monkey. |
| 12:34.20 | alphanet | with some "begin" tags |
| 12:34.29 | alphanet | to tell the script where to store it |
| 12:34.43 | [TK]D-Fender | alphanet: I'd love to see what you've come up with for this if it isn't too "secret". |
| 12:34.46 | alphanet | and yes there are templates |
| 12:34.53 | alphanet | just a moment |
| 12:36.00 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 12:36.10 | *** join/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:36.10 | *** mode/#asterisk [+o lmadsen] by ChanServ |
| 12:39.49 | alphanet | have a nice afternoon |
| 12:45.16 | *** join/#asterisk remibemol (n=remibemo@81-66-205-202.rev.numericable.fr) |
| 12:49.30 | DarnoQ | anyone knows how to convert g729 to wav? |
| 12:51.26 | [TK]D-Fender | DarnoQ: digi.com has an on-line converter, or there is a CLI command if you have licenses already |
| 12:54.29 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 12:54.33 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 12:55.07 | hsv-al | hello all- are we looking forward to another "glorious" week of droning/work |
| 12:55.23 | hsv-al | vacation is over.........staring at monitors and coffee returns.... |
| 12:55.45 | UnixDog | hsv-al: go back to bed .... its to early on a monday |
| 12:55.49 | hsv-al | :) |
| 12:56.06 | hsv-al | awwww |
| 12:56.08 | hsv-al | are your eyes burning |
| 12:56.16 | hsv-al | coffee hasnt "woke" you up yet? |
| 12:56.24 | UnixDog | I have to work on fixing 1.6 issues on bsd this week |
| 12:56.39 | UnixDog | thats bad enough |
| 12:56.54 | hsv-al | hot, i havent touched it yet, only thing I have planned so far is |
| 12:56.58 | hsv-al | mexicanfood + margaritas after work :) |
| 12:57.06 | UnixDog | lol |
| 12:57.58 | hsv-al | unixdog, arent you looking forward to people asking you what good sites there are |
| 12:58.01 | *** join/#asterisk servettas (n=usta@88.249.71.190) |
| 12:58.01 | hsv-al | for open ser configs etc? :( |
| 12:58.02 | DarnoQ | [TK]D-Fender thank you, what is the cli command ? |
| 12:58.13 | CVirus | I just configured an FXO card to receive a call and answer it with the Welcome menu .. but the sound is very low .. suggestions ? |
| 12:58.20 | servettas | hi everyone, i need howto for Open source G.729 and G.723.1 codecs, can anyone hlep me? thanks.. |
| 12:58.31 | UnixDog | I have yet to play with open ser |
| 12:59.11 | UnixDog | g729 and g723 are not open codecs |
| 12:59.30 | UnixDog | g729 cost money 10 dollars per from digium |
| 12:59.31 | *** join/#asterisk SwK (n=SwK@user-24-236-97-87.knology.net) |
| 12:59.33 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:59.46 | servettas | UnixDog, http://asterisk.hosting.lv i mean it sorry |
| 13:00.01 | SwK | does AGI SAY DIGITS not respect the channel's language setting? |
| 13:01.00 | UnixDog | that site is not for us users.. as in the us you have to pay for g729 |
| 13:01.28 | servettas | i can do it |
| 13:01.42 | SwK | that site is not for 90% of the world... |
| 13:01.42 | servettas | but i do not know how can i compile it |
| 13:01.43 | *** join/#asterisk raytruz` (n=raytruz_@96.28.43.212) |
| 13:02.06 | SwK | G729 and G723 are patented and you need to get a properly licensed copy of it |
| 13:02.09 | servettas | i used digium 729a codec |
| 13:02.36 | UnixDog | us patients only count for the US other wise in the rest of the world they look at it as a should be free codec |
| 13:02.48 | SwK | its not just a us patent |
| 13:02.59 | *** join/#asterisk ArashHemmat (n=arash@91.184.77.51) |
| 13:03.10 | SwK | and its not a software patent |
| 13:03.31 | SwK | its a method patent that can be implemented in software or hardware |
| 13:03.49 | SwK | anyway... anyone... agi say digits... setting channel language? |
| 13:04.26 | *** join/#asterisk DrAk0 (n=ljd@nelug/coreteam/luisjose) |
| 13:08.24 | SwK | well its not just AGI its dialplan SayDigits also |
| 13:08.44 | SwK | and yes spanish language files are installed... and its 1.4 latest from svn |
| 13:10.02 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:11.44 | gfather1 | guys i was taking wwith <alphanet> , and im askinh about , like i have 7 phone line from the phone provider to our company |
| 13:12.03 | gfather1 | and i want to connect them to the Asterisk server |
| 13:12.20 | gfather1 | and make the phones in the company , ip phones |
| 13:12.31 | gfather1 | how should the connection be made ? |
| 13:12.40 | gr0mit | gfather1, which country? |
| 13:12.45 | *** join/#asterisk ToTo (n=ToTo@209.8.41.139) |
| 13:12.48 | gfather1 | jordan |
| 13:12.56 | gfather1 | <gr0mit> jordan |
| 13:13.10 | gr0mit | ok so use an 8-port FXO box |
| 13:13.43 | gr0mit | from FXO to SIP, or better, get your telco to migrate the 8 lines to four ISDN BRI |
| 13:13.46 | *** join/#asterisk PodMan99a (n=PodMan99@78-86-189-73.zone2.bethere.co.uk) |
| 13:13.50 | gfather1 | <gr0mit> 8-port FXO box , any link pleas |
| 13:13.51 | gr0mit | and use a quad-bri dcard |
| 13:14.05 | gfather1 | becouse im still learning about asterisk |
| 13:14.08 | gr0mit | never used them, gfather1 |
| 13:14.23 | PodMan99a | hey all... i can make incoming and outgoing calls through SIP on asterisk (Hosted by me off-site) however cannot make internal calls?? any ideas? |
| 13:14.26 | gr0mit | ok well if you are still learning now is not the time to spends lots |
| 13:14.40 | gr0mit | so get a single fxo card and start playing first |
| 13:14.52 | gr0mit | but you are generally much better off using ISDN |
| 13:16.09 | *** join/#asterisk SuPrSluG (n=SuPrSluG@24.75.47.130) |
| 13:16.17 | gfather1 | <gr0mit> : |
| 13:16.28 | gfather1 | :( |
| 13:16.36 | gfather1 | becouse i want to know what parts i should get |
| 13:16.39 | gfather1 | to do the connections |
| 13:16.42 | gfather1 | u know |
| 13:19.07 | *** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com) |
| 13:20.07 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.228.111) |
| 13:23.54 | gfather1 | <gr0mit> http://www.digium.com/en/products/analog/tdm800p.php |
| 13:23.59 | gfather1 | is taht what i need |
| 13:24.00 | gfather1 | ? |
| 13:24.31 | gr0mit | gfather1, yup. |
| 13:24.34 | gr0mit | should be ok |
| 13:24.45 | gfather1 | coolz |
| 13:24.49 | gr0mit | you need to get 8 x fxo modules |
| 13:24.58 | gfather1 | so anything similar to specification like that should be ok |
| 13:25.16 | gfather1 | i tried to search google |
| 13:25.52 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 13:27.09 | PodMan99a | hey all... i can make incoming and outgoing calls through SIP on asterisk (Hosted by me off-site) however cannot make internal calls?? any ideas? |
| 13:27.11 | gfather1 | https://www.8774e4voip.com/SearchResults.asp?Cat=26 |
| 13:27.22 | gfather1 | <gr0mit> see this link pleas |
| 13:28.18 | gr0mit | and? |
| 13:28.42 | gfather1 | there like 20 models |
| 13:28.49 | gfather1 | and they are fxo fx |
| 13:28.52 | gfather1 | :S |
| 13:29.01 | gr0mit | well my recommendation is start cheap |
| 13:29.08 | gfather1 | yes |
| 13:29.10 | gfather1 | me too |
| 13:29.14 | *** join/#asterisk Peri (n=redanti@CPE0012171ab654-CM00080d9b7243.cpe.net.cable.rogers.com) |
| 13:29.15 | gfather1 | but im confuced |
| 13:29.19 | gfather1 | thats the problem |
| 13:29.20 | Peri | morning |
| 13:29.26 | gfather1 | morning |
| 13:29.59 | gfather1 | <gr0mit> im very good at networking and vpn's |
| 13:30.17 | gr0mit | soooo, buy a cheaper card with a single fxo and a single fxs |
| 13:30.18 | gfather1 | but im getting confuced becouse of the analog telephony system |
| 13:30.27 | gr0mit | evaluate, learn and experiment |
| 13:30.35 | gr0mit | then you will know what you need. |
| 13:31.00 | gr0mit | even an old X100P card (ebay/google) is a good starting point |
| 13:31.28 | gr0mit | you can get these for $20 or so. they are horrid but should start you on the learing process |
| 13:33.13 | *** join/#asterisk ToTo (n=ToTo@209.8.41.34) |
| 13:33.36 | gfather1 | <gr0mit> u mean like an 56k modem , right ? |
| 13:33.49 | gfather1 | to test on and stuff |
| 13:34.08 | gr0mit | well, you need to buy the right card |
| 13:34.25 | gr0mit | google for X100P and make sure it says it is compatible iwth Asterisk |
| 13:34.25 | CVirus | What's so bad about the X100P ? |
| 13:34.38 | UnixDog | its a winmodem |
| 13:34.44 | SwK | and they suck |
| 13:34.44 | gr0mit | does not support caller id |
| 13:34.47 | UnixDog | 90% are crap |
| 13:34.51 | gr0mit | fixed impedance |
| 13:34.54 | gr0mit | etc etc |
| 13:34.55 | CVirus | I'm using the X100P Special Edition |
| 13:35.04 | SwK | hah ie: a knock off |
| 13:35.07 | gr0mit | if you are in the USA they are probably fine |
| 13:35.25 | gr0mit | for the rest of the world they are sub-optimal |
| 13:35.55 | SwK | so... anyone running a multi lingual install of 1.4 and actually u sing SayDigits to speak in oh say english and spanish? |
| 13:36.12 | SwK | gr0mit, they are sub-optimal for the entire planet |
| 13:36.30 | gfather1 | is there like any online shop i can search for similar products ? |
| 13:36.34 | UnixDog | you have to install the other lang sound files |
| 13:36.43 | gfather1 | so i can know whats the diffrence and price range and stuff |
| 13:36.44 | SwK | UnixDog, yeah i did that |
| 13:36.48 | gr0mit | SwK, i only know they are suboptimal here in UK |
| 13:36.58 | SwK | gfather1, voip supply |
| 13:36.58 | gr0mit | I can't comment for the rest of the world ;-) |
| 13:37.25 | SwK | gr0mit, i live < 10km from digium and I'll say they suck every where hah |
| 13:37.35 | UnixDog | by default the main sound files that come with asterisk are either english os spanish |
| 13:37.56 | SwK | UnixDog, when you install spanish it installs into sounds/es |
| 13:38.01 | SwK | which is what its supposed to do |
| 13:38.04 | gr0mit | hehe!!!! you must be in Alabama then? |
| 13:38.11 | gfather1 | whats the category for the fxo's ? |
| 13:38.14 | SwK | yeah |
| 13:38.20 | SwK | gfather1, 1 FXO == 1 line |
| 13:38.28 | gfather1 | i know that |
| 13:38.36 | gfather1 | whats the category , |
| 13:38.46 | SwK | gfather1, so it depends on what size card you get... you can get them in current supported configs from 4 to 24 ports |
| 13:38.54 | SwK | they'll be in alaog hardware |
| 13:38.58 | SwK | analog |
| 13:39.05 | gfather1 | ah ok , analog hardware |
| 13:39.34 | SwK | if you need more then say 4 ports its worth it to just spend the few extra dollars get a T1 card and a channel bank tho |
| 13:39.39 | CVirus | what happened to voip-info ? |
| 13:39.46 | CVirus | looks like a new theme or somethign |
| 13:39.49 | CVirus | something* |
| 13:39.58 | gfather1 | http://www.voipsupply.com/index.php?cPath=96_117 |
| 13:40.03 | SwK | wiki- upgrade it loosk like |
| 13:40.06 | gfather1 | these are 1 port fxo , right |
| 13:40.16 | gfather1 | im understanding right |
| 13:40.19 | gfather1 | ? |
| 13:40.23 | gr0mit | channelbanks are a very USA thing |
| 13:40.29 | SwK | those looks like FXS VoIP adapters |
| 13:40.37 | SwK | gr0mit, not really |
| 13:40.49 | gr0mit | they dont really exist anywhere outside USA |
| 13:41.04 | SwK | you just have to know where to look for them ;) |
| 13:41.06 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 13:41.12 | SwK | you can get E1 channel banks |
| 13:41.17 | gr0mit | coz outside usa we all use ISDN2 and ISDN30 |
| 13:41.28 | gr0mit | but we don;t use E1 CAS here |
| 13:41.36 | gr0mit | its all ISDN |
| 13:41.41 | SwK | still a similar principal |
| 13:41.49 | SwK | its just not a "channel bank" |
| 13:41.52 | gr0mit | here an E1 channel bank = PABX ;-) |
| 13:41.58 | SwK | you need something like an atlas 550 |
| 13:42.00 | WildPikachu | gr0mit, still running strong here btw with that callprogress issue :( |
| 13:42.10 | gr0mit | WildPikachu, really?? |
| 13:42.20 | SwK | no pabx required thatway heh |
| 13:42.23 | gr0mit | you mean it is fixed? |
| 13:42.25 | WildPikachu | gr0mit, yep, no dropped calls yet |
| 13:42.28 | *** join/#asterisk J4zen (n=Jeroen@a82-95-153-17.adsl.xs4all.nl) |
| 13:42.29 | WildPikachu | gr0mit, yep |
| 13:42.34 | gr0mit | is baffled! |
| 13:42.40 | SwK | thats how you take a PRI and break it it BRIs (and yes we do that here too) |
| 13:42.56 | gr0mit | SwK, we don;t really ever do that |
| 13:43.15 | SwK | we have mass stupidity here still tho |
| 13:43.22 | SwK | things like GR303 are all the rage still |
| 13:43.23 | gr0mit | again, that function is done by a pbx |
| 13:44.07 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 13:44.23 | SwK | not nessecarily there are times where you dont want to go thru the whole pbx thing... |
| 13:44.24 | gr0mit | SwK, all nasty U-law stuff ;-) |
| 13:44.39 | SwK | and thats where something like the atlas comes into play |
| 13:44.39 | gr0mit | things are much simpler in A-lawland |
| 13:45.17 | SwK | gr0mit, atleast you dont have to deal with our national call routing hah |
| 13:45.21 | gr0mit | (exxept in A-law MFC-R2 land) |
| 13:45.48 | gr0mit | has 2 boxes running R2 signalling |
| 13:45.52 | *** join/#asterisk nicoAMG (i=asgalt@216.25.160.214) |
| 13:47.49 | *** join/#asterisk ZarBla (n=J0ff@modemcable119.221-56-74.mc.videotron.ca) |
| 13:48.04 | ZarBla | <PROTECTED> |
| 13:51.05 | *** join/#asterisk Moudmen (n=Julien@74.50.102.54) |
| 13:51.22 | *** join/#asterisk bboschman (n=bboschma@p50997436.dip0.t-ipconnect.de) |
| 13:51.50 | *** join/#asterisk jainr (n=jainr@65.244.37.75) |
| 13:54.06 | beek | ZarBla: Have you followed the steps for adjusting tx/rx gain on both lines? |
| 13:55.07 | Rico29 | does anyone uses Linksys spa962 phones ? |
| 13:55.26 | gfather1 | so whats the spesification im looking for |
| 13:55.27 | ZarBla | beek: Well I did change it but in the zapata.conf but it doesnt seem to read that config file because I changed hidecallerid to yes in the samefile but I still have callerid... |
| 13:55.30 | gfather1 | as i unserstand |
| 13:55.36 | gfather1 | usa is not like others |
| 13:55.37 | *** join/#asterisk putnopvut (n=putnopvu@216.207.245.1) |
| 13:55.37 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:55.47 | gfather1 | so every country has its diffrences |
| 13:55.58 | gfather1 | but i think jordan is like europe |
| 13:56.35 | *** join/#asterisk fnordus (n=dnall@70.71.224.2) |
| 13:56.36 | *** join/#asterisk tinkerghost (n=eric@host-64-179-18-177.spr.choiceone.net) |
| 13:56.36 | gfather1 | so untill now i dont know what im looking for |
| 13:57.11 | *** join/#asterisk ArashHemmat (n=arash@91.184.88.210) |
| 13:57.11 | gfather1 | guys im really a good listener |
| 13:57.15 | gfather1 | :( |
| 13:57.22 | gfather1 | just need some directions |
| 13:57.22 | hi365 | Running ztcfg: ZT_SPANCONFIG failed on span 1: Invalid argument (22) <---- getting this error, "lsmod|grep ztdummy" show nothing |
| 13:57.28 | beek | ZarBla: search the archives for "milliwatt" test and how to properly configure the gain. You have two lines, so that will make it easier. |
| 13:57.42 | tinkerghost | OK mayday mayday mayday, I am going down ....... My asterisk server isn't transfering anyone to queues & is restarting the queue & message logs 5-10 times a minutes |
| 13:58.12 | tinkerghost | I have already restarted the software & the server itself |
| 13:58.36 | bboschman | Hi |
| 13:58.46 | ZarBla | beek: Ok thanks. Also, my asterisk server (plugged with 2 pots in 2 fxo of my digium) called me like 8 times during the last 11 hours... By itself... Did you see that happen before? |
| 13:58.51 | CVirus | Let's assume that I defined an IVR menu that executes when I call line no. 1 which is connected to my first FXO card ... can I have an IVR option to get the dial one of line no.2 which is connected to my second FXO card ? |
| 13:58.56 | CVirus | is this possible ? |
| 13:59.16 | bboschman | is asterisk multithreaded (or in other words does it makes sense to buy a multicore CPU for asterisk)? |
| 13:59.20 | tinkerghost | when I first logged in this morning asterisk -r logged in but woudln't respond to anything - even exit gave me a blank look |
| 14:00.05 | tinkerghost | bboschman: even if it's not multithreaded, at 2 cores is good because of background tasks |
| 14:00.18 | beek | ZarBla: no, I haven't. Are you sure that you're calling yourself? I'd do an asterisk -rvvvvvvv on a console, core set verbose 9, and then watch the output to see what's going on. |
| 14:00.37 | bboschman | tinkerghost, I'm thinking of buying quadcore :) |
| 14:01.36 | beek | CVirus: DISA |
| 14:03.19 | CVirus | DISA (Direct Inward System Access) |
| 14:03.20 | CVirus | thanks alot :-) |
| 14:03.56 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 14:04.14 | jaytee | morning everyone |
| 14:06.23 | bkruse | sup sup |
| 14:06.43 | jaytee | another wonderful monday by the looks of it. |
| 14:08.39 | MatBoy | jaytee: ow, the end of money is already in sight here |
| 14:09.07 | [TK]D-Fender | CVirus: Don't need DISA. |
| 14:09.30 | [TK]D-Fender | CVirus: you can just do Dial(Zap/g1) or whatever channel/group you want |
| 14:14.01 | jaytee | MatBoy, sorry to hear about your financial woes :-) |
| 14:18.56 | *** join/#asterisk Moudmen (n=Julien@74.50.102.54) |
| 14:19.15 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:21.05 | *** join/#asterisk ArashHemmat (n=arash@91.184.77.24) |
| 14:21.23 | *** join/#asterisk ^shark_ (n=jochieng@41.222.2.65) |
| 14:22.12 | *** join/#asterisk LakeSolon (n=blake@64-83-198-152.dhcp.stcd.mn.charter.com) |
| 14:22.21 | CVirus | [TK]D-Fender: cool ... thanks alot |
| 14:24.36 | CVirus | I want an option in the IVR menu for speed dialing and saving or deleting items from the speed dial menu ... what to look for ? |
| 14:25.38 | jblack | oh, probably either odbc, or agi. |
| 14:26.06 | Moudmen | i'm having a problem with my a2billing callback, whenever the 1st leg hangs up, a2billing charges the user only for the 1st leg rate, and considers that the 2nd leg didn't answer. but everything works fine when the 2nd leg hangs up. does anybody have any idea about this ? |
| 14:28.39 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 14:28.44 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.138) |
| 14:29.31 | *** join/#asterisk servettas (n=usta@88.249.71.190) |
| 14:29.42 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:30.20 | *** join/#asterisk idc-dutch (n=idc-dutc@195-144-082-203.dyn.adsl.xs4all.be) |
| 14:31.51 | *** join/#asterisk spokra (n=spokra@host093-179-142.sea0.speakeasy.net) |
| 14:31.53 | idc-dutch | hello room, I have a problem with asterisk 1.4 where sip extension keep hinted HOLD when a call is put on hold, and is disconnected. Any known issue? |
| 14:31.56 | MatBoy | jaytee: I ment monday :D |
| 14:32.07 | MatBoy | LOL |
| 14:32.24 | jaytee | MatBoy, I know......I was joking! :-) |
| 14:32.25 | MatBoy | and his typos |
| 14:32.32 | MatBoy | jaytee: :P |
| 14:32.41 | MatBoy | man, now my links are down again |
| 14:32.45 | MatBoy | and don come up |
| 14:32.55 | jaytee | there's always too much month left at the end of the money. |
| 14:33.10 | MatBoy | jaytee: that is a good one ! |
| 14:33.29 | idc-dutch | ~centos52bug |
| 14:33.30 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 14:34.09 | MatBoy | mhh, this kinda sucks |
| 14:34.19 | CVirus | isn't there an application that can draw asterisk dial plans ? |
| 14:34.24 | CVirus | a diagram I mean |
| 14:34.30 | MatBoy | ow that would be nice |
| 14:35.27 | *** join/#asterisk ToTo (n=ToTo@207.176.6.208) |
| 14:37.25 | MatBoy | jaytee: do you ever used misdn |
| 14:37.28 | MatBoy | <PROTECTED> |
| 14:37.42 | jaytee | MatBoy, nope |
| 14:37.48 | CVirus | jblack: any other suggestions ? |
| 14:38.24 | jblack | Yeah. Brush your teeth every morning and every night, to have good food choices when you get old |
| 14:39.27 | CVirus | was my question stupid to that extent ? |
| 14:39.45 | jblack | Kinda. |
| 14:39.55 | viraptor | I've got an MeetMe app running with latest high precision ztdummy (min. 99,98% on zttest) and asterisk running on high priority, but I still hear late packets / pauses sometimes - what else can I try to improve the quality? |
| 14:40.38 | jblack | First, you singled someone out that didn't indicate any interest in the subject. Second, the person you singled out thinks that any dialplan front end is going to be awkward and less flexible than spending the week it takes to get a good grasp of dialplans. |
| 14:42.55 | *** join/#asterisk jpcansa (n=jpbenavi@201.201.66.155) |
| 14:44.29 | *** join/#asterisk casix (n=casix@pbxedifici.adamvozip.es) |
| 14:44.40 | casix | hello |
| 14:44.53 | *** part/#asterisk ^shark_ (n=jochieng@41.222.2.65) |
| 14:46.20 | casix | anyone knows wich differences are between the errors: "Everyone is busy/congested" and "SIP/channel is circuit-busy"?? |
| 14:46.54 | *** join/#asterisk seanmh (i=seanmh@216.31.101.25) |
| 14:47.52 | jblack | THat's a good question. I don't have an answer that doesn't involve rationalization |
| 14:52.30 | casix | po vale |
| 14:53.42 | casix | but is circuit-busy is only that the destination is busy? that it cannot create the channel because there is no route to destination? or what? |
| 14:57.14 | *** join/#asterisk af_ (n=getsmart@88-149-230-202.dynamic.ngi.it) |
| 15:02.08 | *** join/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 15:02.30 | EricL | Which scheduler is best to use with Asterisk (or does it not really make a difference)? |
| 15:04.42 | *** join/#asterisk shtoom (n=shtoom@121.246.167.147) |
| 15:05.14 | jblack | realtime |
| 15:06.17 | EricL | jblack: I mean the I/O scheduler. CFQ, deadline, or anticipatory? |
| 15:07.08 | jblack | no idea then. |
| 15:07.39 | jblack | * is generally not io limited, unless you're trying to record 20 calls at the same time |
| 15:16.12 | *** join/#asterisk hfb (n=hfb@pool-71-118-254-245.lsanca.dsl-w.verizon.net) |
| 15:22.09 | *** join/#asterisk freezey (i=hidden-u@gw.mypublisher.com) |
| 15:22.11 | freezey | join #linux |
| 15:23.10 | *** join/#asterisk chandoo (n=chandra@ool-4353b9e5.dyn.optonline.net) |
| 15:23.29 | *** join/#asterisk shido6 (n=shido6@209.114.208.192) |
| 15:28.14 | *** join/#asterisk outtolunc (n=me@c-67-170-211-86.hsd1.ca.comcast.net) |
| 15:29.43 | EricL | jblack: Thanks. |
| 15:29.45 | *** part/#asterisk EricL (n=eric@jarbeeg.chal.net) |
| 15:31.51 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 15:35.43 | *** join/#asterisk oilinki3 (n=oil@ppp-124-120-17-184.revip2.asianet.co.th) |
| 15:36.23 | *** join/#asterisk jeremy_g (n=jerms@static-213-115-44-90.sme.bredbandsbolaget.se) |
| 15:37.15 | jeremy_g | my asterisk proxy needs to register with a remote proxy using a certain display name. how do i do that in sip.conf? should i set calleridname=desired displayname |
| 15:43.21 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 15:44.07 | *** join/#asterisk PepOSX (n=angeldav@200.90.100.98) |
| 15:44.36 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.static.dsl.tele.dk) |
| 15:46.54 | jeremy_g | so dead |
| 15:51.03 | *** join/#asterisk bkw_ (n=brian@adsl-70-234-182-25.dsl.tul2ok.sbcglobal.net) |
| 15:52.15 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 15:53.15 | *** part/#asterisk jarod14 (n=jarod14@LMontsouris-152-63-1-19.w80-12.abo.wanadoo.fr) |
| 15:53.18 | *** join/#asterisk gfather1 (n=enforcer@79.173.197.94) |
| 15:53.24 | gfather1 | hello guys |
| 15:53.41 | gfather1 | i understand now what is the fxs and fxo |
| 15:53.48 | gfather1 | :) |
| 15:53.55 | gfather1 | so to make my connection |
| 15:54.28 | gfather1 | i need an fxs/fxo card or device to connect to the ip/pbx wich is the asterisk |
| 15:55.01 | gfather1 | then the office phones will be ip based |
| 15:55.04 | gfather1 | right? |
| 15:55.48 | gfather1 | is it better to get an fxs/fxo device or a pci card ? |
| 15:57.28 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 15:57.30 | spokra | gfather1: how many lines are you looking at? |
| 15:57.55 | gfather1 | i have 7 phone lines from the phone company |
| 15:58.30 | gfather1 | and like lets say 12 inline phones inside the company |
| 15:58.37 | gfather1 | for local if u get what i mean |
| 15:58.56 | spokra | didgiam has a 8 port pci card do you need any analog phone off your pbx? say for a fax machine |
| 15:59.22 | gfather1 | yes |
| 15:59.25 | *** join/#asterisk ManxPower (n=manxpowe@238.sub-75-248-137.myvzw.com) |
| 15:59.44 | gfather1 | like the fax will not be connected to the ip/pbx right |
| 15:59.56 | gfather1 | or the fxs/fxo card |
| 15:59.59 | gfather1 | or im wronge |
| 16:00.01 | gfather1 | ? |
| 16:00.01 | spokra | it could be if you wanted it to |
| 16:00.17 | ManxPower | Generally, it is best not to run fax thru Asterisk |
| 16:00.28 | gfather1 | i see |
| 16:00.32 | *** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca) |
| 16:00.39 | gfather1 | is there any good site for shopping |
| 16:00.52 | gfather1 | im looking , but still dont know what are best sites for such stuff |
| 16:00.55 | ManxPower | If you can, just connect the fax machine direct to the telco line. You CAN run fax thru Asterisk, but it will not be as reliable as doing it direct. |
| 16:01.06 | gfather1 | yes i see |
| 16:01.32 | ManxPower | gfather1: Fax has much more strict requirements than voice. |
| 16:01.38 | spokra | http://www.digium.com/en/ and pick a vendor from there |
| 16:02.09 | gfather1 | <spokra> in the store section |
| 16:02.25 | spokra | link at the top of the page where to buy |
| 16:02.31 | gfather1 | <ManxPower> im thinking to connect it directly , or to the fxs/fxo card |
| 16:04.35 | gfather1 | there is no seller in jordan |
| 16:04.39 | gfather1 | only in dubai :) |
| 16:04.45 | gfather1 | but its still in the region |
| 16:05.30 | *** join/#asterisk XnOSX (n=XnOSX@212.145.172.127) |
| 16:08.28 | *** join/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 16:08.29 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 16:17.28 | *** join/#asterisk Che-Anarc (i=Che-Anar@bb-82-108-13-47.ukonline.co.uk) |
| 16:18.48 | Che-Anarc | I want to see if I can get a basic Asterisk server up using my voice modem... and am looking at "Howtos and Tutorials" on http://www.voip-info.org/wiki/index.php?page=Asterisk can anyone recommened a link that is good for newb's / first-timers trying this out? |
| 16:19.18 | Peri | check out the stuff on asteriskguru.com |
| 16:19.28 | Peri | they've got a lot of very nice tutorials |
| 16:19.37 | jeremy_g | my asterisk proxy needs to register with a remote proxy using a certain display name. how do i do that in sip.conf? should i set calleridname=desired displayname |
| 16:19.53 | Che-Anarc | Peri thanks |
| 16:20.08 | jeremy_g | ManxPower, i expect you to say sth about thart |
| 16:21.59 | gfather1 | guys for a 50 local phones , as an example |
| 16:22.14 | gfather1 | what should the server requirments be |
| 16:22.34 | ManxPower | Che-Anarc: Asterisk does not support voicemodems. At one point it tried, but that support was removed long ago because it did not work |
| 16:23.06 | gfather1 | and what does the term ( multeple server in distrebuted architecture ) means ? |
| 16:25.01 | ManxPower | jeremy_g: usually you would want to do a Set(CALLERID(name)=Robert Dobbs) or similar just before you Dial the remote side. |
| 16:25.04 | jblack | Would anyone know offhand if it's possible to get polycoms to use primary and backup * servers? |
| 16:25.39 | ManxPower | jblack: according to the Polycom Admin Docs, you can set primary and backup SIP servers. I've never done it, never heard of anyone trying to do it. |
| 16:25.55 | jblack | Ok. I'll hunt those down |
| 16:26.41 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 16:26.41 | *** mode/#asterisk [+o russellb] by ChanServ |
| 16:27.32 | [TK]D-Fender | jblack: it'll try the secodary if the primary failes IIRC. Not sure if it'll switch cold on first failure, or if it will do it on a call/call basis |
| 16:27.49 | jblack | wow, that's easy. |
| 16:28.00 | jblack | voipprotserver.x.address |
| 16:29.15 | gfather1 | guys any one can answear me pleas |
| 16:29.16 | gfather1 | ? |
| 16:29.30 | jblack | I wish there were polycoms that could do 802.11g |
| 16:29.40 | jblack | I'm probably go broke buying them |
| 16:29.46 | jblack | s/I'm/I'd |
| 16:31.18 | jeremy_g | ManxPower:yeah i know that but its registeration. |
| 16:31.31 | jeremy_g | [TK]D-Fender:any thoughts on my problem |
| 16:31.40 | jeremy_g | my asterisk proxy needs to register with a remote proxy using a certain display name. how do i do that in sip.conf? should i set calleridname=desired displayname |
| 16:31.56 | Peri | [TK]D-Fender and jblack, it will do it on a call by call basis |
| 16:32.27 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-182-234-rndf-esr-5.dynamic.isadsl.co.za) |
| 16:32.58 | [TK]D-Fender | jeremy_g: *'s "register =>" sucks and has little configurability. You're probably screwed. |
| 16:34.36 | Kobaz | jblack: i wish there were polycoms that didn't take several minutes to boot |
| 16:35.37 | Peri | jeremy_g do you need the CID on register, or on call? |
| 16:35.53 | Peri | Kobaz i'm with you on that |
| 16:36.10 | gfather1 | what should the server requirments be for more the 30 local phones ? |
| 16:36.45 | *** join/#asterisk orionr (n=orionr@c-76-26-221-76.hsd1.sc.comcast.net) |
| 16:38.01 | gfather1 | and what rg5 cabels should be used |
| 16:38.42 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:39.22 | gfather1 | like im gonna connect the server to a 50 port switch |
| 16:39.48 | gfather1 | the cabel between the server and the switch can handel the data ? |
| 16:41.58 | ManxPower | It requires callerid on registration? Best of luck with that. |
| 16:42.24 | ManxPower | Kobaz: I hope you don't have to reboot your polcoms very often |
| 16:42.50 | ManxPower | gfather1: you will fail at Asterisk if you don't understand networking. |
| 16:43.48 | gfather1 | i understand networking |
| 16:44.04 | ManxPower | Then why did you ask what kind of cable to use? |
| 16:44.06 | gfather1 | but there are like 10 mb cables |
| 16:44.13 | ManxPower | not really. |
| 16:44.22 | gfather1 | ah sorry , its teh ports not the cabels |
| 16:44.25 | ManxPower | the only 10Mb cables are cat3 and have not been around for many many years. |
| 16:44.44 | ManxPower | ports? Use more words so we understand you. |
| 16:44.47 | [TK]D-Fender | s/many/decade |
| 16:45.03 | gfather1 | there are switches who have 10 mb ports |
| 16:45.14 | ManxPower | gfather1: not in many years |
| 16:45.18 | gfather1 | and there are 100 mb ports |
| 16:45.21 | ManxPower | Unless you are using very old equipment |
| 16:45.26 | gfather1 | <ManxPower> where i live there is man |
| 16:45.30 | [TK]D-Fender | 10 mb SWITCH? I've only seen HUBS at this point,. |
| 16:45.31 | gfather1 | no |
| 16:45.42 | gfather1 | well allot of chines stuff gets here |
| 16:45.56 | gfather1 | and u see a normal 4 port switch with 10 mb speed |
| 16:46.03 | ManxPower | gfather1: one G711 alaw call will use .080 Mbps. |
| 16:46.15 | ManxPower | now, how many calls could you send via 10 Mbps? |
| 16:46.26 | gfather1 | alot |
| 16:46.38 | ManxPower | exactly. So why are you worried about it? |
| 16:46.52 | *** join/#asterisk scampbell (n=scampbel@199.105.195.156) |
| 16:46.52 | gfather1 | nah not the calls |
| 16:47.05 | gfather1 | but wouldent it get latency or something |
| 16:47.18 | ManxPower | gfather1: not unless the link was saturated. |
| 16:47.18 | gfather1 | with allot of connections on same port ? |
| 16:47.25 | gfather1 | ah oks |
| 16:47.29 | gfather1 | good |
| 16:47.34 | ManxPower | gfather1: Asterisk can handle 32,000 connections on one port. |
| 16:47.40 | *** join/#asterisk Kyoshi (i=whoa@pool-71-167-117-15.nycmny.fios.verizon.net) |
| 16:47.45 | gfather1 | very impressive:) |
| 16:48.00 | gfather1 | whats the system requirments for local 50 phones |
| 16:48.05 | ManxPower | gfather1: not really, it's the same limit most software has that requires 2 ports per connection. |
| 16:48.17 | ManxPower | gfather1: nobody can tell you that because there are too many variables. |
| 16:48.52 | gfather1 | mmmm depends on what |
| 16:49.02 | gfather1 | so i can like now what should i buy |
| 16:49.10 | ManxPower | codec usage, transcodinf, recording, logging level, disk activity, interupt latency |
| 16:49.28 | Kyoshi | When installing asterisk 1.4.21, i want to use unixODBC, so i install version 2.2.12, but MenuSelect does not recognize ODBC therefore I cannot satisfy the requirements to install res_odbc and other components. THis has been driving me nuts for a few days now and the info I found on voip-info is not helping much, Any input would be greatly appreciated. |
| 16:49.31 | spokra | maybe an appliance would work.. http://www.digium.com/en/products/appliance/?tab=features |
| 16:49.53 | ManxPower | gfather1: most modern computers could handle 50 users without too much of a problem, unless you are using G729 or other high CPU setting |
| 16:49.54 | Qwell | Kyoshi: Do you have the odbc devel packages? |
| 16:49.55 | gfather1 | <ManxPower> then atleast i have to know what minemum should i get? |
| 16:50.14 | gfather1 | ah i see |
| 16:50.21 | Kyoshi | qwell, yes, been installed |
| 16:50.35 | Kyoshi | i yum'd all required packaged |
| 16:50.36 | ManxPower | gfather1: Min for most settings with 50 users: 2Ghz Pentium 4, 1GB of RAM |
| 16:50.38 | Qwell | did you re-run configure? |
| 16:50.43 | Kyoshi | yes i did |
| 16:50.49 | Qwell | and what does it say? |
| 16:50.55 | Kyoshi | make clean && ./configure && make menuselect |
| 16:51.00 | gfather1 | ManxPower> very good :) |
| 16:51.09 | gfather1 | so i know what system i should get |
| 16:51.26 | gfather1 | so a new pc with price of like 800 $ should do the deal |
| 16:51.28 | Kyoshi | say? about odbc? |
| 16:51.37 | Qwell | yes |
| 16:51.42 | ManxPower | gfather1: I am assuming, of course, that only a small number of those 50 phones will be in use at any one time, assuming you are not transcoding to/from G729 and are not using call recording. |
| 16:51.53 | Kyoshi | lemme check |
| 16:52.39 | *** join/#asterisk svenna_ (n=svenna@p548D164F.dip0.t-ipconnect.de) |
| 16:52.48 | ManxPower | gfather1: usually with Asterisk you should disable the build in network port and use an ethernet card in the server. |
| 16:53.09 | Qwell | ManxPower: why? O.o |
| 16:53.50 | ManxPower | Qwell: Built in 100Gbs motherboard networking frequently locks interrupts for a long time causing issues with Asterisk and Zaptel |
| 16:54.15 | ManxPower | Putting in an ethernet card is cheap, easy way to solve the issue. |
| 16:54.32 | twisted | uhhh |
| 16:54.35 | ManxPower | sorry, that would be 1000Mbps, not 100Gbps |
| 16:54.43 | Qwell | bugzilla.kernel.org... |
| 16:54.50 | ManxPower | twisted: it has solved the issue for many people |
| 16:55.02 | twisted | really.. hmm... |
| 16:55.22 | ManxPower | twisted: at least solved it for enough people spending $30 on a network card is cheap insurance |
| 16:55.29 | twisted | true |
| 16:55.56 | twisted | it's gotta be mfr related or something... i haven't had but one system thus far who has experienced anything like that |
| 16:56.12 | *** join/#asterisk gfather (n=enforcer@79.173.212.166) |
| 16:56.19 | ManxPower | We use all 3com cards, except for the 1000Mbs 3Com card, which is really a Broadcom card and didn't even work well enough for our e-mail server to keep connections up. |
| 16:56.20 | gfather | sorry i was disconnected |
| 16:56.35 | ManxPower | twisted: I assume it's a chipset thing, but as I said, it's cheap insurance. |
| 16:56.40 | twisted | nods |
| 16:56.55 | gfather | <ManxPower> so now i know what should i get |
| 16:57.04 | gfather | :) |
| 16:57.22 | Kyoshi | qwell, i logged the results from configure, what should i be looking for? |
| 16:57.36 | twisted | & |
| 16:57.45 | ManxPower | twisted: onboard SATA stuff does it too. At least modern Digium cards are much, much, much less dependent on interrupt latency, so the issue is less common these days |
| 16:57.56 | Qwell | Kyoshi: anything about odbc.. |
| 16:58.13 | Kyoshi | got ya |
| 16:58.26 | ManxPower | "HDLC Abort" is the symptom of this IRQ latency problem when using a Zaptel card. |
| 16:58.44 | ManxPower | Kyoshi: try putting the ./configure output on pastebin.ca |
| 16:58.54 | ManxPower | so Qwell can actually SEE what you see. |
| 16:59.30 | *** join/#asterisk servettas (n=usta@88.249.71.190) |
| 17:00.35 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 17:00.39 | casix | bye |
| 17:00.43 | servettas | i have a problem about asterisk server http://paste.ubuntu.com/25722/ can anyone help me ? |
| 17:01.15 | ManxPower | servettas: the message is quite clear |
| 17:01.20 | *** join/#asterisk fskrotzki (n=fskrotzk@host.textwise.com) |
| 17:01.47 | servettas | ManxPower, [Jul 7 16:50:58] WARNING[3191] codec_g729a.c: out of G.729 decoder licenses |
| 17:01.55 | servettas | this is error messages |
| 17:02.09 | ManxPower | what part of that do you not understand. you have used all your licenses -- buy more. |
| 17:02.25 | servettas | but just buyed |
| 17:02.32 | [TK]D-Fender | servettas: NOT ENOUGH |
| 17:02.41 | ManxPower | how many licences did you buy? |
| 17:02.43 | servettas | must i buy more |
| 17:02.49 | jeev | HEY FINDER! |
| 17:02.50 | jeev | i mean fender |
| 17:02.51 | jeev | :d |
| 17:03.28 | [TK]D-Fender | servettas: You're system needed more than you had at that point in time. |
| 17:03.43 | [TK]D-Fender | servettas: If those circumstances occur often enough, its your call |
| 17:03.44 | ManxPower | servettas how many licences did you buy? I won't ask a 3rd time. |
| 17:03.51 | servettas | ManxPower, http://paste.ubuntu.com/25724/ |
| 17:03.57 | servettas | here my codec |
| 17:04.25 | ManxPower | servettas: either answer my question or I will stop helping you. |
| 17:04.38 | ManxPower | show translations does NOT show your configuration. |
| 17:04.39 | bkruse | ManxPower: I am not sure exactly what me means by here my codec, lol |
| 17:04.56 | hi365 | ztcfg -vvvv is returning: ZT_SPANCONFIG failed on span 1: Invalid argument (22) any ideas why? http://pastebin.ca/1064778 |
| 17:05.00 | bkruse | if he has the real g729, it is g729 show licenses I believe |
| 17:05.12 | bkruse | hi365: what does zttool look like? |
| 17:05.19 | ManxPower | bkruse: or he could just remember how many he bought |
| 17:05.20 | servettas | ManxPower, how can i see it |
| 17:05.22 | servettas | ? |
| 17:05.23 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:05.32 | bkruse | ManxPower: You should let this one go....lol |
| 17:05.34 | hi365 | bkruse: zttool: command not found |
| 17:05.35 | ManxPower | servettas: Do you not remember how many licenses you purchased? |
| 17:05.38 | bkruse | servettas: type 'g729 show licenses' |
| 17:05.41 | ManxPower | bkruse: I shall. |
| 17:05.44 | bkruse | hi365: you've got a problem.... |
| 17:05.58 | ManxPower | servettas: I cannot help you further. Perhaps bkruse can help you. Best of luck. |
| 17:06.23 | blinky42 | Polycom + Asterisk question - is there a way to have First & Last name show up on the phones when you dial an extension instead of just the extension number? Would that be on the asterisk side in the sip call setup, or in the phones? |
| 17:06.32 | hi365 | bkruse: no kidding! :) i belive i installed all the defaults. how so i start debuging? |
| 17:06.37 | bkruse | servettas: type that command, and what is the number after the words "licensed channels available" |
| 17:06.37 | ManxPower | blinky42: Not at this time. |
| 17:06.40 | [TK]D-Fender | blinky42: Not currently |
| 17:06.43 | bkruse | hi365: did you install from source? |
| 17:06.50 | hi365 | yes |
| 17:06.52 | ManxPower | blinky42: people might say you can, but they are confused. |
| 17:07.03 | bkruse | hi365: go into zaptel and type 'make install' or ./zttool |
| 17:07.12 | [TK]D-Fender | blinky42: there is a patch for CPID (Called Party ID) out there, but its not merged into 1.4 or lower. I'm not sure if any of it made it into 1.6 series either. |
| 17:07.24 | ManxPower | blinky42: The polycom CAN display the dialed name IF the name/number is in the phone's address book. |
| 17:07.33 | servettas | bkruse, i am using centos and i don't know how can i see it |
| 17:07.34 | blinky42 | oh well. thanks. yea i am running 1.6beta9 |
| 17:07.39 | ManxPower | but that is NOT an Asterisk thing, that is a phone things |
| 17:07.56 | Kyoshi | qwell,the only reference to odbc found from ./configure is "checking for SQLConnect in -lodbc... yes |
| 17:07.58 | bkruse | servettas: asterisk -rx 'g729 show licenses' |
| 17:08.12 | blinky42 | ManxPower, i was thnking that and just loaded up the local phone book on 2 phones but it doesn't seem to display it |
| 17:08.14 | ManxPower | bkruse: welcome to our world. 8-| |
| 17:08.22 | servettas | asterisk -rx 'g729 show licenses' |
| 17:08.22 | servettas | No such command 'g729 show' (type 'help' for help) |
| 17:08.32 | ManxPower | blinky42: there is a config option. Read the admin manual |
| 17:08.35 | hi365 | bkruse: ran make install - nothing, [root@myhost zaptel-1.4.11]# ./zttool |
| 17:08.35 | hi365 | -bash: ./zttool: No such file or directory |
| 17:08.50 | bkruse | servettas: asterisk -r; g729 <tab><tab> |
| 17:09.04 | bkruse | ManxPower: if he says g729 <tab><tab> command not found, banhammer. lol |
| 17:09.22 | hi365 | bkruse: i seem to have a dependency issue with zttool |
| 17:09.30 | bkruse | then it did not build because libnewt |
| 17:09.34 | Qwell | servettas: How did you obtain the codec, and the licenses? |
| 17:09.47 | hi365 | bkruse: correct. ill try to install it |
| 17:10.27 | ManxPower | FOUR of some of the the most experienced Asterisk people in the world and they all are having problems helping servettas. I wonder what the common thing is...... |
| 17:10.30 | servettas | Qwell, from digium |
| 17:10.52 | hardwire | can you use a sourceip (for routing reasons) with sip/iax peers? |
| 17:10.55 | bkruse | Qwell: Feel free to +b when you want.... |
| 17:10.58 | hardwire | in sip.conf and iax.conf? |
| 17:11.04 | hardwire | different per peer. |
| 17:11.13 | [TK]D-Fender | bkruse: Not warranted yet. |
| 17:11.16 | bkruse | hi365: Awesome, what zttool will tell you is what configuration the card is currently in (if any) and how many channels are available. What is likely happening is that when you run ztcfg -vv and it tries to set a config on a channel that does not exist (eg this happens when you try to load e1 settings on a card in t1 mode) |
| 17:11.30 | hardwire | I should be able to set the TOS per peer on a privileged machine. |
| 17:11.38 | bkruse | servettas: You are saying you got your g729 licenses from qwell? |
| 17:11.44 | bkruse | Qwell: Doing some side business? :P |
| 17:11.59 | [TK]D-Fender | bkruse: Context comprehension failure :) |
| 17:12.00 | servettas | bkruse, i got from digium |
| 17:12.09 | servettas | :( |
| 17:12.21 | ManxPower | I'll bet he installed the codec, but not the license |
| 17:12.31 | ManxPower | servettas: How much did you PAY FOR YOUR LICENSES? |
| 17:12.39 | Qwell | servettas: Please contact Digium technical support. |
| 17:12.39 | servettas | 10$ |
| 17:13.00 | ManxPower | servettas: then you have ONE license. Many things in Asterisk require 2 license. |
| 17:13.06 | [TK]D-Fender | servettas: Taht will allos 1 transcoded call on your system. |
| 17:13.07 | servettas | ok |
| 17:13.22 | ManxPower | hands [TK]D-Fender more coffee |
| 17:13.28 | servettas | ManxPower, thanks for your help and thanks to everybody for all |
| 17:13.33 | [TK]D-Fender | servettas: Next time you get that error, pastebin "sip show channels". |
| 17:13.34 | bkruse | servettas: If you bought them from Digium, Digium will give you support :) |
| 17:13.40 | servettas | i will buy one or more |
| 17:14.00 | bkruse | servettas: np, the only problem is that you are probably needing 2 channels to transcode more than one call, or a complex call, record, etc |
| 17:14.17 | servettas | i understand |
| 17:14.19 | servettas | 194.221.62.198 irisoptik 0683b7ef1c8 00131/00000 0x0 (nothing) No |
| 17:14.19 | servettas | 1 active SIP channel |
| 17:14.39 | servettas | now it mean i have onlye one licence |
| 17:14.45 | [TK]D-Fender | servettas: And "core show channels concise"? |
| 17:15.35 | servettas | core show channels [concise|verbose] |
| 17:15.35 | servettas | <PROTECTED> |
| 17:15.35 | servettas | <PROTECTED> |
| 17:15.42 | *** join/#asterisk jarod14 (n=jarod14@ns1.viatelecom.com) |
| 17:15.52 | [TK]D-Fender | serSo no active channels? |
| 17:16.07 | *** part/#asterisk ManxPower (n=manxpowe@238.sub-75-248-137.myvzw.com) |
| 17:16.09 | [TK]D-Fender | servettas: ^^ |
| 17:16.16 | servettas | nothing looking |
| 17:16.33 | [TK]D-Fender | servettas: then you have not properly set up your codec at all. |
| 17:16.53 | servettas | what i must do now ? |
| 17:16.57 | outtolunc | you would think since the translation matrix for g729 to g729 is '-' on his system, that would be a clue |
| 17:17.14 | [TK]D-Fender | servettas: Because with no channels in use its not a question of not having enough licenses.... you don't have ANY functional yet. Go read the install procedures and if this still fails, call Digium support |
| 17:17.14 | servettas | ok i understand |
| 17:17.31 | [TK]D-Fender | outtolunc: Perhaps the codec is ok, but the license file load is not. |
| 17:17.49 | outtolunc | exactly, only 1 leg |
| 17:18.00 | servettas | ok |
| 17:18.02 | [TK]D-Fender | outtolunc: No, I mean *0* loaded due to config problem. |
| 17:18.03 | servettas | thanks for all |
| 17:18.08 | servettas | i will try now |
| 17:18.11 | [TK]D-Fender | outtolunc: He can't even start 1 leg or so it seems. |
| 17:19.33 | outtolunc | ah |
| 17:20.37 | *** join/#asterisk [hC] (n=hardcore@190.10.9.126) |
| 17:25.44 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 17:26.03 | *** join/#asterisk vetoni (n=vetoni@pool-71-123-209-104.dllstx.fios.verizon.net) |
| 17:31.47 | Kyoshi | qwell? |
| 17:33.55 | *** join/#asterisk jets (n=brian@pdpc/supporter/active/jets) |
| 17:35.33 | *** join/#asterisk mvanbaak (n=michiel@vanbaak.xs4all.nl) |
| 17:40.48 | kamanashisroy | hi, I installed asterisk GUI from CVS .. configured manager.conf and http.conf .. But when I try to login, it says login successful and then it shows the login screen again .. |
| 17:42.03 | DarnoQ | try restarting the server and flushing browser cache |
| 17:42.07 | *** join/#asterisk Zochwar (i=noone@062016241070.customer.alfanett.no) |
| 17:42.14 | bkruse | kamanashisroy: flush browser cache, what revision of the gui? |
| 17:42.17 | bkruse | btw #asterisk-gui |
| 17:42.40 | kamanashisroy | bkruse: sorry |
| 17:44.11 | bkruse | kamanashisroy: np |
| 17:45.10 | *** join/#asterisk sacitec (n=tobi@201.144.211.82) |
| 17:45.29 | sacitec | ~centos52bug |
| 17:45.30 | jbot | [~centos52bug] There is a bug compiling Zaptel "xpp" (Astribank) driver with CentOS/RHEL 5.2. Either uncheck it from MENUCONFIG, or please read the following for instructions on how to correct : http://bugs.digium.com/view.php?id=12889 |
| 17:46.14 | *** join/#asterisk cvox (n=chatzill@c-71-195-192-97.hsd1.ut.comcast.net) |
| 17:46.36 | Zochwar | Can someone with a little more experience than me recommend a driver for a regular HFC PCI ISDN card? I'm testing mISDN now, but it seems really unstable (is the zaptel driver better, google found a visdn driver, but that seems really outdated). Running latest linux kernel. |
| 17:47.57 | *** join/#asterisk l2trace99 (n=asd@static-71-251-65-16.tampfl.fios.verizon.net) |
| 17:48.47 | *** join/#asterisk hsv-al (n=hsval@66.0.46.210) |
| 17:48.48 | hsv-al | . |
| 17:48.55 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 17:49.53 | hi365 | bkruse: onme more thing: it doesnt seem like that change is permanent. how do i make it stik? |
| 17:50.44 | hi365 | stick |
| 17:50.52 | hsv-al | ugh |
| 17:51.38 | bkruse | hi365: um....the BEST way to do it is to change the jumper on the card |
| 17:51.54 | hi365 | is reading the manual to figure out how to do that |
| 17:52.00 | Kyoshi | When installing asterisk 1.4.21, i want to use unixODBC, so i install version 2.2.12, but MenuSelect does not recognize ODBC therefore I cannot satisfy the requirements to install res_odbc and other components. THis has been driving me nuts for a few days now and the info I found on voip-info is not helping much, Any input would be greatly appreciated. |
| 17:53.27 | bkruse | hi365: I am not sure where that is, i cannot remember |
| 17:53.51 | hsv-al | bkruse, did you play CS all weekend? |
| 17:54.01 | [TK]D-Fender | Kyoshi: re-so "./configure" |
| 17:54.04 | [TK]D-Fender | re-do |
| 17:54.17 | bkruse | hi365: http://www.asteriskguru.com/tutorials/wildcard_te405p_wildcard_te410p.html |
| 17:54.24 | bkruse | hsv-al: I played gcc all weekend |
| 17:54.35 | hsv-al | heh |
| 17:54.41 | bkruse | hi365: search for "jumper" it shows closed and open, similar on the te120p that you have |
| 17:54.52 | Kyoshi | tkd: already did a few times... "make clean && ./configure && make menuselect" and still all ODBC related items are X'd out |
| 17:55.06 | hi365 | bkruse:problem is with jumper = e1 and mina had the jumper on the whole time :( |
| 17:55.33 | hi365 | is reading here: http://www.google.co.il/url?sa=t&ct=res&cd=1&url=http%3A%2F%2Fwww.olantel.com%2Fdoc%2Fte120-series-manual.pdf&ei=XFdySIDlJJW40gWyl-zmAQ&usg=AFQjCNHn8JD5zXfDOn3K6iW7wJs9Ti6eaA&sig2=3B8psKLj6PW017AO-drbhg |
| 17:55.46 | hi365 | ops, www.olantel.com/doc/te120-series-manual.pdf |
| 17:56.55 | bkruse | hi365: the other document I gave you is easier. Search for the name "jumper" |
| 17:56.55 | sacitec | hello, i'm have 2 asterisk (1.4.19) boxes linked via sip trunk but still without communication. One box is behind NAT with cisco 2801 router. I already declared a static route to UDP 5060 and opened 10000- 20000 port range for RTP but still i'm unable to get them wo work, any clue what i'm missing ? |
| 17:57.19 | [TK]D-Fender | sacitec: Plenty of SIP.CONF settings to ensure. Read up : |
| 17:57.21 | [TK]D-Fender | ~sipnat |
| 17:57.21 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 17:57.23 | [TK]D-Fender | ^^^^^^^^^^^^^^^^ |
| 17:57.38 | Kyoshi | [TK]D-Fender: I already did a few times... "make clean && ./configure && make menuselect" and still all ODBC related items are X'd out |
| 17:57.56 | hi365 | bkruse: funny, now it works. i guess the jumper is defective :) |
| 17:58.15 | *** join/#asterisk kannan (n=kann@121.246.224.66) |
| 17:58.47 | [TK]D-Fender | Kyoshi: Ah.. and make sure to re-run ldconfig so taht your .so mappings are solid |
| 17:59.17 | Kyoshi | [TK]D-Fender: ldconfig? not familiar with that |
| 17:59.31 | [TK]D-Fender | Kyoshi: go run it as-is |
| 17:59.39 | Kyoshi | okie dokie |
| 17:59.41 | Kyoshi | done |
| 17:59.47 | Kyoshi | reconfigure is all then? |
| 18:00.11 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:00.16 | kannan | hi to all.This is the first time i am using FXO card in asterisk.I do all the things when i am dial from asterik cli the call answered.but my phone not ringing. |
| 18:00.19 | [TK]D-Fender | Kyoshi: should just need "ldconfig", then re-extract, ./configure , make imenuconfig |
| 18:00.44 | [TK]D-Fender | kannan: pastebin is your friend, and FFS don't dial from CLI, use at least a softphone. |
| 18:00.46 | Kyoshi | so trash my prior extraction, will do |
| 18:01.01 | [TK]D-Fender | Kyoshi: Always best for a guaranteed clean start |
| 18:01.07 | Kyoshi | so true |
| 18:01.48 | kannan | D-Fender:thank you.I try and get back to u. |
| 18:06.18 | hi365 | wow digium has good documentation! |
| 18:07.09 | *** join/#asterisk jameswf-home (n=james@dsl093-157-131.phx1.dsl.speakeasy.net) |
| 18:09.24 | kannan | D-Fender:Facing same problem |
| 18:09.34 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:10.07 | kannan | D-Fender:Below I paste the output |
| 18:10.11 | kannan | Executing AGI("SIP/2002-081eba60", "call_log.agi|99391116312") in new stack |
| 18:10.12 | kannan | <PROTECTED> |
| 18:10.12 | kannan | <PROTECTED> |
| 18:10.12 | kannan | <PROTECTED> |
| 18:10.13 | kannan | <PROTECTED> |
| 18:10.15 | kannan | <PROTECTED> |
| 18:10.17 | kannan | <PROTECTED> |
| 18:10.19 | kannan | <PROTECTED> |
| 18:10.21 | kannan | <PROTECTED> |
| 18:10.22 | Kobaz | yaaay a flood |
| 18:10.24 | kannan | Jul 7 23:38:53 WARNING[3957]: res_agi.c:210 launch_netscript: Connect to 'agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----28-----25)' failed: Connection refused |
| 18:10.27 | kannan | <PROTECTED> |
| 18:10.33 | M1s3ry | kannan, pastebin |
| 18:10.41 | Kobaz | kannan: don't do that |
| 18:10.53 | jameswf-home | ~pb |
| 18:10.54 | jbot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 18:11.13 | M1s3ry | HA |
| 18:11.31 | M1s3ry | wow.... |
| 18:12.50 | *** join/#asterisk kannan (n=kann@121.246.224.66) |
| 18:14.23 | *** join/#asterisk lanning (n=lanning@ip67-152-85-190.z85-152-67.customer.algx.net) |
| 18:14.27 | kannan | hi, I am having TDM2400p FXO card. |
| 18:14.45 | *** join/#asterisk ipstatic (n=ipstatic@24.106.202.78) |
| 18:15.14 | ipstatic | Has anyone here remapped a softkey on a Polycom phone? |
| 18:16.20 | kannan | anyone help on TDM2400p card |
| 18:16.22 | jameswf-home | no but i did stay at a holiday in express last night |
| 18:17.00 | hi365 | shh, your wife might find out |
| 18:17.21 | jameswf-home | find out? she made me share :( |
| 18:17.41 | hi365 | :) |
| 18:17.53 | j0 | anyone here setup a pri with telus? |
| 18:18.41 | jpcansa | i got a probem with my dialout code, http://pastebin.com/d7fdebeae |
| 18:20.05 | kannan | pastebin:hi |
| 18:20.37 | jameswf-home | jpcansa: ${AVAILCHAN} << not a valid parameter |
| 18:20.44 | jameswf-home | *not populated |
| 18:21.42 | jpcansa | jameswf: it doesnt happend all the time, why it is not populated sometimes? |
| 18:22.37 | seanbright | jpcansa: because none of the channels you pass to ChanIsAvail are available? |
| 18:23.18 | jpcansa | thats not right, theres available channels |
| 18:23.20 | [TK]D-Fender | kannan: Looks like it dialed jsut fine |
| 18:23.34 | seanbright | jpcansa: ok. |
| 18:23.46 | seanbright | jpcansa: except that you're wrong. |
| 18:24.01 | jameswf-home | if the channel is unavailible it may not populate the variable add a exten => _XXXXXXXX,999,NOOP(FAIL) |
| 18:24.08 | seanbright | jpcansa: if ${AVAILCHAN} is empty, it means that none of the channels you pass to ChanIsAvail are available. |
| 18:24.35 | vetoni | how many simultaneous SIP/IAX registrations does * allow per extension? is there a way to limit it? |
| 18:25.06 | [TK]D-Fender | vetoni: only 1 device can be registered to a given account entry at a time. |
| 18:25.20 | vetoni | thx |
| 18:25.24 | jpcansa | seanbright: as soon as i got the error, i check core show channels and there are available channels |
| 18:25.35 | jameswf-home | jots down seanbright's response " ok, BUT YOUR WRONG" for use at a later time |
| 18:25.43 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 18:26.01 | seanbright | jameswf-home: i didn't capitalize. and i spelled "you're" correctly ;-) |
| 18:26.09 | jameswf-home | bastard.. |
| 18:26.11 | jameswf-home | :) |
| 18:26.32 | hi365 | bkruse: thansk again. have a good night |
| 18:26.44 | [TK]D-Fender | jpcansa: just dial eash group back-to-back directly and forget about "chanisavail". It isn't needed. |
| 18:26.54 | [TK]D-Fender | each* |
| 18:27.15 | seanbright | jpcansa: ok. but that doesn't change the fact that when you call ChanIsAvail and it sets ${AVAILCHAN} to empty, none of the channels that you passed to it are available. |
| 18:27.58 | jameswf-home | jpcansa: NOOP is your friend... everywhere you use a variable drop a noop after it is set and you can see wht they are set to |
| 18:28.28 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 18:29.40 | jpcansa | [TK]D-Fender: what do you mean bac-to-back directly? |
| 18:30.04 | *** join/#asterisk simprix (n=simprix@cosmas.supportdept.com) |
| 18:30.07 | jpcansa | jameswf: good idea |
| 18:30.27 | *** join/#asterisk cesar_CR (n=cesar@200.91.75.45) |
| 18:31.26 | *** part/#asterisk vetoni (n=vetoni@pool-71-123-209-104.dllstx.fios.verizon.net) |
| 18:32.41 | jameswf-home | jpcansa: Dial(Zap/g1&Zap/g2&Zap/g3/${EXTEN}) |
| 18:34.12 | *** join/#asterisk tyler__ (n=tyler@65.46.5.122) |
| 18:34.25 | tyler__ | Can you now have 2 different extentions? When I add 2, the first one stops working |
| 18:34.31 | tyler__ | rasterisk says its not found |
| 18:34.32 | tyler__ | when I call |
| 18:34.41 | jpcansa | jameswf: that´ll check for available channel? |
| 18:38.26 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 18:42.37 | jameswf-home | tyler__: ask better |
| 18:46.56 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 18:52.03 | *** join/#asterisk mchou (n=mchou@c-76-103-44-118.hsd1.ca.comcast.net) |
| 18:52.31 | mchou | anyone here have experience with VoiceXML? |
| 18:52.58 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 18:55.02 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 18:57.26 | *** join/#asterisk kannan (n=kann@121.246.224.66) |
| 18:58.14 | macros73 | anyone have experience with GXP2000 Grandstream phones? Darn thing is refusing to upgrade via TFTP. |
| 19:01.32 | kannan | hi to all,I am newly using the TDM2400P card in asterisk for going using the FCT(mobile).zap will configure.(channel 01: FXS Kewlstart (Default) (Slaves: 01)) it show like this when using ztcfg -vv.when I connect the phone line it show alarm recieved on Zap 18.We need to connect all the channels or any one of the channels? |
| 19:02.40 | *** join/#asterisk aspendora (n=chatzill@adsl-68-92-55-163.dsl.hstntx.swbell.net) |
| 19:06.44 | [TK]D-Fender | kannan: Go call Digium for installation support |
| 19:08.19 | macros73 | ok, wireshark is my friend. This phone is calling for files not included in the freaking firmware update. |
| 19:09.00 | kannan | D-Fender:from here i am not able to call |
| 19:10.58 | jeev | Fender, so far.. i dunno what it was.. moving the asterisk server from the datacenter to here, or having my wholesaler move me to the newer asterisk server.. no static, NO PROBLEMS.. PERFECT++ |
| 19:11.53 | jaytee | sometimes you come across hardware that really drives you crazy, be it a laptop, desktop or server you'd rather just take it to some field someplace and take a sledgehammer to it while listening to Damn it feels good to be a gangsta. |
| 19:12.39 | *** join/#asterisk [hC] (n=hardcore@201.204.2.226) |
| 19:13.12 | [TK]D-Fender | kannan: Why not? |
| 19:13.34 | [TK]D-Fender | jeev: ok, fine, sure. |
| 19:13.35 | seanbright | because his TDM2400P doesn't work |
| 19:13.36 | seanbright | ;) |
| 19:13.49 | jaytee | macros73, what files in particular? I've used SolarWinds TFTP server to update firmware. |
| 19:13.55 | [TK]D-Fender | seanbright: Nope.. not an excuse yet :) |
| 19:14.05 | [TK]D-Fender | ~gs |
| 19:14.05 | jbot | GrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice. |
| 19:14.30 | macros73 | jaytee: gxp2000a.bin and boot55a.bin |
| 19:14.50 | jaytee | macros73, hang on a sec |
| 19:14.58 | macros73 | jaytee: I had to go WAY back to a copy of 1.1.2 before I found firmware updates with those file names...guess I need ot update with it first, then try again |
| 19:17.41 | jaytee | macros73, I downloaded their latest about 2 months ago and I'm running 1.1.5.15 for software and bootloader is 1.1.5.6 and I've got both those files in the TFTP root. |
| 19:18.10 | kannan | D-Fender:I am from India.And also in remote Place.If i need to plug the power to this card or not? |
| 19:19.10 | [TK]D-Fender | kannan: you can call Digum via IAX2. |
| 19:19.11 | jaytee | did you download the firmware updates from them? |
| 19:19.21 | macros73 | jaytee: I was trying to update to 1.1.6.16 |
| 19:19.37 | macros73 | jaytee: Yes, the 1.1.6 version I did. Had to get 1.1.2 from Grandstreamsucks. |
| 19:19.49 | macros73 | jaytee: It doesn't look like gs makes earlier versions available on their website |
| 19:20.08 | jaytee | I'm pretty sure they do but the link isn't obvious |
| 19:20.27 | *** join/#asterisk fedya (n=fedya@rrcs-71-43-222-2.se.biz.rr.com) |
| 19:21.28 | kannan | D-Fender:ok.Thank you |
| 19:22.50 | macros73 | jaytee: Now that i've updated to 1.1.2.25, it's asking for a boot55b.bin...which also isn't in 1.1.6. :P |
| 19:23.10 | macros73 | jaytee: I'll check for a 1.1.5 on gssucks |
| 19:23.31 | *** join/#asterisk bsaxon (n=bsaxon@119.sub-75-248-116.myvzw.com) |
| 19:30.12 | Peri | yep |
| 19:31.02 | *** join/#asterisk hi365_m (n=hi365@213.151.56.96) |
| 19:31.45 | jaytee | macros73, I have the 1.1.5.15 in a zip file. I could email it to you if you're email doesn't block large attachments. |
| 19:32.04 | jaytee | it's 10.2MB |
| 19:32.28 | macros73 | Shouldn't be a problem, msg'd you my email. |
| 19:32.45 | macros73 | and thanks |
| 19:38.13 | *** join/#asterisk Segnale007 (n=Segnale0@host115-10-dynamic.33-79-r.retail.telecomitalia.it) |
| 19:40.43 | jaytee | macros73, ok it's on it's way. I used my gmail account too so it should be there quick or already be there. |
| 19:43.38 | macros73 | jaytee: Weird. The fonts look cleaner in the updated firmware. |
| 19:44.25 | Kyoshi | qwell, d-fender, your advise together hellped, thanks so much. |
| 19:45.40 | [TK]D-Fender | Kyoshi: Glad to help |
| 19:58.08 | Qwell | Kyoshi: and what was the fix? |
| 19:59.29 | [TK]D-Fender | Qwell: ldconfig to prep the linking I'm betting |
| 20:00.24 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 20:01.21 | jaytee | macros73, not only that but if you turn off silence suppression it actually works. It would say it was but it never actually did it in the earlier 1.1.2.x firmware. |
| 20:01.30 | jaytee | that would play havoc with the MOH. |
| 20:02.03 | macros73 | jaytee: lol. What can I do about echoing or smearing? I've tried turning down the mic gain in the new firmware, no improvement. Also set the tx per frame to 1. |
| 20:02.45 | jaytee | macros73, I left those settings at their defaults. Are you using TDM cards? |
| 20:03.21 | macros73 | jaytee: Nope, straight SIP out to the ITSP. Of course, the other end is a PRI. |
| 20:03.27 | macros73 | diff phone system, though |
| 20:04.26 | jaytee | because SIP phones will get jitter but echo usually is from the bridged circuit not the SIP phone and you can't adjust for echo in sip. |
| 20:05.36 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:05.37 | macros73 | jaytee: Weird part is, if I call the same person via Ekiga through the same Asterisk box, the callee says I sound "fine." |
| 20:05.53 | *** join/#asterisk NightKhaos (n=nightkha@78-86-111-126.zone2.bethere.co.uk) |
| 20:05.55 | macros73 | jaytee: But if I call via the GXP2000, I am "blurry." |
| 20:06.00 | NightKhaos | Hi there. |
| 20:06.56 | *** join/#asterisk Nugget (i=nugget@carrera.macnugget.org) |
| 20:07.10 | NightKhaos | I want to setup a rather simple setup, connecting my server to the landline via a telephony card and then allowing SIP devices to make VoIP calls to my server via my landline. But I need some help, first and formost, what card should I get (I'm running gentoo) and where from (i'm in the UK)? |
| 20:08.24 | andrewy | NightKhaos: most cards for connecting to a landline are quite expensive or have problems (echo, etc). you're probably best off with something like a sipura 3000, which connects to the landline and your network, and allows asterisk to access the landline through it via SIP |
| 20:08.29 | [TK]D-Fender | NightKhaos: Digium TDM-410P with 1 FXO module, or a Linksys SPA-3102. |
| 20:08.57 | macros73 | Ah no, don't answer before D-Fender, you'll make him mad and he'll ridicule us. |
| 20:09.04 | [TK]D-Fender | NightKhaos: Where from would be whatever retailer you can find locally to offer the best price. |
| 20:09.45 | [TK]D-Fender | macros73 : Don't be bitter over getting burnt by that Gradstream of yours ;) |
| 20:10.05 | lesouvage | Does anybody have experience with using asterisk in combination with CheckPoint firewall. Is this a combination that can work well. Any suggestion is very welcome. |
| 20:10.07 | NightKhaos | andrewy: if I was going to do that, why would I require Asterisk? I would simply redirect my phones to use the Sipura? |
| 20:10.39 | macros73 | Bah. Grandstream. |
| 20:11.22 | [TK]D-Fender | NightKhaos : Noone said you required Asterisk at all. |
| 20:11.25 | andrewy | NightKhaos: true, sip phones could place a call without asterisk |
| 20:11.55 | macros73 | oooh, wait. I can make a PoE injector from the GXP's power supply. |
| 20:12.21 | [TK]D-Fender | NightKhaos: And SIP devices don't call your server via your landline. The might call your server via SIP to ACCESS your landline however |
| 20:12.54 | NightKhaos | [TK]D-Fender: now now, no need to get pandatic. |
| 20:13.14 | lesouvage | NightKhaos: If this is a low budget project and just for testing and trying you could also consider a x100p card. It is not perfect, not suitable for use in production, but you can use it to learn and try. |
| 20:13.29 | [TK]D-Fender | NightKhaos: Well when what you say doesn't add up, it'd be nice to be fully sure of your intentions.... |
| 20:13.58 | [TK]D-Fender | NightKhaos: But if my extrapolation of it was on the mark, you've got your answer. |
| 20:14.00 | NightKhaos | [TK]D-Fender: very well... I'll start from the start and try and use better english because I clearly wasn't. |
| 20:14.30 | lesouvage | NightKhaos: you could also considering spending some time reading the book |
| 20:14.42 | [TK]D-Fender | NightKhaos: was just a tiny bit off, but in a possibly crucial way ;) |
| 20:14.51 | lesouvage | ~the book |
| 20:15.12 | Strom_M | ~book |
| 20:15.12 | 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, or see ~buybook |
| 20:15.21 | *** join/#asterisk wonderworld (n=ww@ip-62-143-31-126.hsi.ish.de) |
| 20:16.29 | NightKhaos | Simply put I have preexsisting WiFi infrastructure in my house (2 indepenent APs) and rather than fiddle around with a DECT system and wireless repeaters to get coverage thought my house I thought it would be easier, more cost efficent (considering the size of the house) to the preexsisting WiFi instructure, not to mention this method allows cellphone users in our household to use the landline rather than their (rather vaulable) p |
| 20:17.32 | NightKhaos | Also, since DECT and WiFi are in the same band, possible interfearance was also another consideration. |
| 20:18.15 | [TK]D-Fender | NightKhaos: For consideration : |
| 20:18.18 | [TK]D-Fender | ~wifivoip |
| 20:18.19 | jbot | [~wifivoip] Wi-Fi (802.11a/b/g) VoIP phones on the market may possess any of the following drawbacks : poor battery life, limited range, lack of call features (many lack transfer, etc), poor or no NAT traversal, shoddy construction, etc. Because of this they are usually not recommended. Some alternatives are : VoIP ATA + cordless, DECT w/ VoIP base, etc |
| 20:18.20 | wonderworld | hi, i have trouble setting up asterisk with the allowguest-option=yes. I want people to be able to simply call my IP with a softphone, eliminating the need to register with a sip-provider to contact me. It works, call get thru and my softphones ring when someone calls. but after that we can't hear eachother and the call is hung up by * after a few seconds. Asterisk logs say: "Maximum retries exceeded on transmission"... My * Box is not |
| 20:18.20 | wonderworld | NATed. It works fine when people call via my sip-provider (going thru the same context in extensions.conf) |
| 20:19.42 | NightKhaos | [TK]D-Fender: considered, considered (the modem I use features hotspot handoff features), NAT isn't a problem as it's an internal network, considered. |
| 20:19.47 | [TK]D-Fender | wonderworld: you need to set all the settings (except externip/host) as though you WERE behind NAT so that * knows not to trust their inbound IP. |
| 20:20.00 | NightKhaos | modem? ha/ |
| 20:20.01 | [TK]D-Fender | NightKhaos: Just food for thought. |
| 20:20.02 | NightKhaos | PHONE. |
| 20:20.36 | NightKhaos | [TK]D-Fender: and I thank you, it's good to know someone is making sure I am not getting way over my head. :) |
| 20:20.39 | [TK]D-Fender | NightKhaos: So you've got your 2 best starter options for SPTN interface for * now. |
| 20:20.51 | wonderworld | [TK]D-Fender: thanks, so that would basicly be nat=yes. anything else? |
| 20:21.11 | [TK]D-Fender | wonderworld: canreinvite=no , qualify=yes |
| 20:21.18 | [TK]D-Fender | pstn* |
| 20:22.23 | NightKhaos | [TK]D-Fender: Do I need a * box for this particular configuration? If not, what are the benefits of having one compared to just using the Sipura box? |
| 20:24.43 | [TK]D-Fender | NightKhaos: * can let you do intelligent stuff. You won't need to direct IP's around, you can do VM and the 100's of other things * offers. Hook up to other ITSP's, etc... |
| 20:25.20 | *** join/#asterisk rpm (n=rUssell@69.46.119.121) |
| 20:25.28 | rpm | anyone know if the polycom microbrowser works on polycom 501's? |
| 20:25.48 | [TK]D-Fender | rpm: On SIP 2.2.0+ yes |
| 20:25.55 | [TK]D-Fender | RPM, sorry, 2.1.0 |
| 20:26.11 | wonderworld | [TK]D-Fender: tnx, gonnat try that. didn't know about externip. that must have been the problem |
| 20:26.27 | [TK]D-Fender | wonderworld: it isn't if you're on a public IP. |
| 20:26.46 | wonderworld | i am on dyndns so the right setting would be Asterisk SIP externhost = my.dyndns.org right? |
| 20:26.50 | [TK]D-Fender | wonds just that the OTHER settings need to be taken into consideration for [general] as your calls are coming in un-authed |
| 20:26.56 | Kyoshi | qwell, i needed to YUM alot more than I thought. i was used to ast 1.2 |
| 20:27.00 | NightKhaos | [TK]D-Fender: would you recommend a complete removable of all PSTN direct (i.e. analog) replacing them SIP WiFi and hardline ethernet phones? |
| 20:27.13 | [TK]D-Fender | wonderworld: as you just said you weren't behind NAT, and are now more and more consistently contradicting. |
| 20:27.16 | [TK]D-Fender | wonderworld: read up : |
| 20:27.18 | [TK]D-Fender | ~sipnat |
| 20:27.19 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 20:27.20 | [TK]D-Fender | ^^^^^^^^^^^^^^^^^ |
| 20:27.55 | wonderworld | well my * box is not natted. my clients are. NAT is done by the * box itself for my lan |
| 20:28.15 | [TK]D-Fender | NightKhaos: SIP hardphones are mush more naturally functional, but would I spend on new hardware? Often, especially for home, no. |
| 20:28.34 | [TK]D-Fender | NightKhaos: ATA's work jsut great and let you use the phones you have at minimal expense. |
| 20:28.52 | NightKhaos | [TK]D-Fender: I'm thinking of getting this small starter kit for £106.93, 1x AX100-P Telepony card and 2x AT-530 |
| 20:29.03 | [TK]D-Fender | NightKhaos: and the SPA-3102 already lets you connect up to X analog phones sharing its FXS port |
| 20:29.42 | [TK]D-Fender | NightKhaos: Oh god.... those Atcom phones are shit-on-a-stick. |
| 20:29.51 | NightKhaos | [TK]D-Fender: that is why I asked. |
| 20:30.12 | *** join/#asterisk angeldavid (n=angeldav@nelug/coreteam/pepo) |
| 20:30.14 | [TK]D-Fender | NightKhaos: are your phones US RJ11 compliant by any chance? |
| 20:30.30 | [TK]D-Fender | NightKhaos: or reaadily adaptable? |
| 20:30.53 | NightKhaos | [TK]D-Fender: is that the... erm... 4 pin ethernet style plug? |
| 20:31.03 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:31.19 | macros73 | I bought an Aastra 480iCT with the handset for home use/testing/playing. Very happy with it. |
| 20:31.57 | NightKhaos | [TK]D-Fender: google says yes. :) |
| 20:32.19 | [TK]D-Fender | NightKhaos: Do a little research. The Linksys SPA ATA series is pretty decent. |
| 20:32.26 | [TK]D-Fender | ok, heading home, back in a little bit. |
| 20:33.22 | wonderworld | is there a way to initiate test-calls so i don't have to bother my friend all the time? |
| 20:34.29 | macros73 | wonderworld: Do you only have one friend? :D I make test calls to my cellphone and make the wife answer it if I need a voice on the other end. Or to a receptionist. |
| 20:34.48 | [netman] | wonderworld: AMI |
| 20:34.56 | NightKhaos | Okay so.... is my BT landline a POTS or FXO? I'm confused. :) |
| 20:35.04 | [netman] | exactly, Originate |
| 20:35.17 | wonderworld | well i need to test incoming calls to my ip via sip-phone. all my friends are non-geeks ;) |
| 20:35.35 | lmadsen | I'm having an issue where I need StartMusicOnHold() followed by StopMusicOnHold(), and then followed by StartMusicOnHold() again to always start at the beginning of a MOH file, instead of where the MOH was stopped by the StopMusicOnHold() appilcation -- anyone know if this is expected behaviour? |
| 20:36.56 | macros73 | wonderworld: Are you testing whether the call connects, or call quality? |
| 20:37.25 | wonderworld | i am trying to debug the whole thing. the call gets thru but is dropped after a few seconds without voice being transmitted |
| 20:38.14 | macros73 | wonderworld: Okay. You can call 911 and shout "Oh god no don't kill me", then hang up. They'll try to call you back. When the call keeps getting dropped, they'll dispatch someone who can help you troubleshoot further. :D More seriously, have you checked your log files to see why it's disconnecting? |
| 20:39.45 | wonderworld | yes. i am getting "Maximum retries exceeded on transmission" errors [TK]D-Fender suggested to adjust my NAT settings. i did but i just need someone to test them with now |
| 20:39.56 | wonderworld | i'll go for the 911 aproach |
| 20:40.36 | jblack | wonderworld: Nah. DOn't call 911. |
| 20:40.40 | jblack | What do you want? Sip? |
| 20:40.54 | wonderworld | yes |
| 20:41.03 | jblack | try 0@mercury.linuxguru.net |
| 20:41.15 | wonderworld | no i need incoming |
| 20:41.35 | jblack | I'll help you in 10 if someone else hasn't. |
| 20:41.40 | wonderworld | tnx a lot |
| 20:43.36 | *** join/#asterisk Kyler (n=chatzill@smtp.phaseit.net) |
| 20:43.44 | macros73 | wonderworld: Ack, no, do not call 911. |
| 20:43.45 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 20:44.07 | wonderworld | hehe. i won't |
| 20:44.34 | Kyler | I'm trying to determine if Dictate() hangs up after awhile when paused. It looks like it uses ast_waitstream() but it's not clear to me if that has a timeout. |
| 20:51.04 | *** join/#asterisk mvicha (n=someaddr@190.42.75.126) |
| 20:52.32 | mvicha | hello guys. I wonder if someone could help me mapping some key sequence to do a hook-flash. I'm actually using an addpac telephone which gets the flash button through mgcp, but it's connected to asterisk actually, so I don't have that button working and I need it to do call waiting and 3-way calling. |
| 20:52.34 | mvicha | :s |
| 20:54.18 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:55.36 | *** join/#asterisk jong2 (n=chatzill@65.100.10.89) |
| 20:56.38 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 20:57.55 | mvicha | ne1? |
| 20:58.08 | jblack | wonderworld: still here? |
| 20:58.50 | wonderworld | yes |
| 20:59.05 | wonderworld | my friend is testing again with me |
| 20:59.12 | jblack | Ok. great. |
| 20:59.37 | wonderworld | i debuged the incoming sip requests.... he seems to transmit his internal lan ip so my asterisk can't establish the voice channel |
| 20:59.47 | jblack | try the "nat = yes" option |
| 21:00.03 | jblack | and don't forget to forward the ports you set up in rtp.conf |
| 21:00.45 | wonderworld | my asterisk is not natted |
| 21:00.54 | wonderworld | the box is connected directly to the internet |
| 21:01.06 | wonderworld | having a dynamic ip from dyndns |
| 21:03.21 | wonderworld | this is my sip.conf right now -> http://pastebin.com/m34e65d31 |
| 21:03.48 | *** join/#asterisk suma (n=suma@c-71-196-147-116.hsd1.co.comcast.net) |
| 21:05.43 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:05.54 | *** part/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 21:05.57 | *** join/#asterisk tompaw (n=tompaw@pav.vip.krakow.tompaw.net) |
| 21:06.52 | tompaw | Hi |
| 21:07.19 | tompaw | Which Prepaid Application would you suggest as the best? |
| 21:08.41 | mvicha | hello guys. I wonder if someone could help me mapping some key sequence to do a hook-flash. I'm actually using an addpac telephone which gets the flash button through mgcp, but it's connected to asterisk actually, so I don't have that button working and I need it to do call waiting and 3-way calling. |
| 21:12.34 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
| 21:13.26 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 21:16.51 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 21:18.27 | *** join/#asterisk sysreq` (n=sysreq@unaffiliated/sysreq) |
| 21:22.05 | mvicha | could ne1 at least tell me where may I ask about it? |
| 21:23.37 | Strom_C | mvicha: if you're using an IP phone, there's no need for a hookflash button |
| 21:24.39 | mvicha | Strom_C: how do you do 3 way calling if you don't have one? |
| 21:24.51 | Strom_C | there's usually a button labeled "conference" |
| 21:25.33 | tompaw | is asterisk2billing worth a try? |
| 21:25.40 | *** join/#asterisk jpcansa (n=jpbenavi@201.198.58.30) |
| 21:25.46 | mvicha | well, the problem is that those buttons may not be used. If I would have the addpac gateway I would probably be able to use them, but I need to use addpac |
| 21:26.02 | mvicha | the problem is that those buttons are set through mgcp I think. |
| 21:26.29 | [TK]D-Fender | mvicha: You're using MGCP phones with *? |
| 21:27.37 | mvicha | [TK]D-Fender: Yes, it's a headache for sure. |
| 21:27.55 | mvicha | but I can't use the buttons :s |
| 21:27.57 | [TK]D-Fender | mvicha: What model? |
| 21:28.25 | mvicha | it's addpac ap-ip 100, addpac ap-vp300 |
| 21:29.07 | wonderworld | [TK]D-Fender: it's still me, would you mind to look at my sip.conf to see if anything is right for getting direct calls on my ip? -> http://pastebin.com/m34e65d31 |
| 21:29.14 | [TK]D-Fender | mvicha: ok, well if your phone can't offer it, I'm not sure how * will... |
| 21:30.09 | mvicha | well, if I use the addpac gateway I get the flash button working, but I wonder if I could do some key mapping to flash() for sip |
| 21:30.16 | [TK]D-Fender | wonderworld: "nat=yes", and your register has to come AFTER everything else under [general] |
| 21:30.25 | mvicha | I know it's possible for zap, but it doesn't work on sip channels. |
| 21:30.43 | wonderworld | [TK]D-Fender: ok i'll try, thanks |
| 21:30.50 | [TK]D-Fender | mvicha: "flash" isn't a real VoIP option |
| 21:31.19 | [TK]D-Fender | mvicha: Hate to say it, but good luck with those phones... |
| 21:31.27 | mvicha | LOL |
| 21:32.41 | mvicha | it's a really PITA. You aren't able to get a second call neither, as you can't put your actual call in waiting |
| 21:32.43 | mvicha | :( |
| 21:33.19 | [TK]D-Fender | mvicha: Cut your losses, sell them off and replace them. |
| 21:33.41 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 21:33.51 | *** part/#asterisk Cresl1n (n=matt@216.207.245.1) |
| 21:34.08 | mvicha | yeap, I think that's the best option. Or use it for my grandmom :p |
| 21:34.18 | [T]ank | selling a sangoma a104d (for port t1 card) for $500, any takers? |
| 21:34.28 | [T]ank | few months old. works just fine |
| 21:34.31 | [T]ank | no longer using it |
| 21:34.45 | wonderworld | [TK]D-Fender: still not working. i'll give up for today. thanks for the help though |
| 21:34.46 | mvicha | thanks a lot guys. Will continue with something else |
| 21:35.01 | [TK]D-Fender | wonderworld: you should show your latest config |
| 21:35.08 | [TK]D-Fender | wonderworld: And a failed call with SIP debug |
| 21:35.13 | mvicha | bye bye |
| 21:35.29 | wonderworld | ok |
| 21:36.36 | *** part/#asterisk Rem| (n=remi@bg-fw2out.monmouth.com) |
| 21:38.58 | wonderworld | is there any easy way to log the sip debug into a file? |
| 21:39.26 | tompaw | I attach to that question |
| 21:40.13 | [TK]D-Fender | wonderworld: copy&paste :p |
| 21:40.30 | wonderworld | hehe ok |
| 21:42.46 | *** join/#asterisk bsaxon (n=bsaxon@96.sub-70-220-139.myvzw.com) |
| 21:43.48 | jblack | I believe it's logged into /var/log/asterisk/Master |
| 21:49.39 | wonderworld | ok, here it is -> http://pastebin.com/m2eef4dd6 |
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| 21:51.51 | *** join/#asterisk xacatecas (n=jkroon@dsl-240-135-216.telkomadsl.co.za) |
| 21:52.19 | xacatecas | hi guys, quick question, if I record to g729 when using MixMonitor - what options do I have for playback? |
| 21:52.33 | *** join/#asterisk edwin_quijada (n=macaruch@25.116.88.200.m.sta.codetel.net.do) |
| 21:52.37 | [TK]D-Fender | wonderworld: Too much crap in there. Make sure your ITSP entries all say "nat=no" |
| 21:52.45 | edwin_quijada | how can i know what codec i have installed? |
| 21:52.51 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 21:52.55 | [TK]D-Fender | xacatecas: the same as any other sound file |
| 21:53.07 | [TK]D-Fender | edwin_quijada: "core show modules like codec" |
| 21:53.19 | xacatecas | play it back via a phone ... what can i provide people in terms of downloading the recorded files? |
| 21:53.27 | anonymouz666 | [TK]D-Fender haha "too much crap" was great |
| 21:53.31 | [TK]D-Fender | wonderworld: then retry and trim the exceses and include everything from the START of your call. |
| 21:53.36 | edwin_quijada | [TK]D-Fender: Thks |
| 21:53.52 | [TK]D-Fender | excess* |
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| 21:55.31 | NightKhaos | [TK]D-Fender: in the end I went for a Zaptel based card. |
| 21:55.56 | [TK]D-Fender | NightKhaos: Ok. |
| 21:56.45 | [TK]D-Fender | NightKhaos: but I advise against Zaptel FXS. ATA's are a better idea. |
| 21:56.58 | NightKhaos | [TK]D-Fender: it's just a cheap 1 port telephony card that I'm gonna connect my * server too. |
| 21:57.31 | [TK]D-Fender | NightKhaos: Uh oh... thats sounding more and more like an X100P.... |
| 21:57.34 | Qwell | so, you ignored the advice, and got a crap card... |
| 21:57.57 | *** join/#asterisk Ebola (n=Ebola@unaffiliated/ebola) |
| 21:58.19 | NightKhaos | [TK]D-Fender: to late, order payed for. :) And it wasn't my decision to make. |
| 21:58.38 | Qwell | it's not like it's hard to throw away $20 |
| 21:58.44 | [TK]D-Fender | NightKhaos: Oh? Who's was it? |
| 21:58.54 | NightKhaos | [TK]D-Fender: my father's |
| 21:59.28 | [TK]D-Fender | NightKhaos: Hope it works out for you |
| 22:00.44 | edwin_quijada | [TK]D-Fender: it not working for me |
| 22:01.21 | [TK]D-Fender | edwin_quijada: Please avoid non-descript pronouns.... |
| 22:01.34 | wonderworld | [TK]D-Fender: sorry, i can't today, have no time left. tnx for the help |
| 22:02.19 | NightKhaos | So what is so bad about the X100P anyway? |
| 22:02.40 | edwin_quijada | [TK]D-Fender: ok, was module show like codec |
| 22:02.47 | edwin_quijada | [TK]D-Fender: thks! |
| 22:02.51 | [netman] | NightKhaos: it's not reliable |
| 22:03.51 | NightKhaos | [netman]: okay... you're gonna have to be more specific. Cause I'm getting one, weither I like it or not. |
| 22:04.54 | [netman] | I would only use a X100P to test and play, never for professional use |
| 22:05.06 | [TK]D-Fender | NightKhaos: Poor CID detection and call disconnect support functionality. Shoddy PCI interface, etc. |
| 22:05.36 | [netman] | a lot of headaches |
| 22:06.33 | NightKhaos | [TK]D-Fender, [netman]: key words here: home use. |
| 22:07.04 | [netman] | absolutely, NightKhaos |
| 22:07.12 | [TK]D-Fender | NightKhaos: And when it fails to release yoru home line on remote diconnect your home line might not be able to take calls for a while.. |
| 22:08.17 | j0 | does a pri need special setup to get outbound callerid working? i'm using a sangoma card if that makes a difference |
| 22:10.06 | [TK]D-Fender | j0: No. Your telco does have to permit it however |
| 22:10.26 | NightKhaos | [TK]D-Fender: that was very vandictive, and not at all helpful. |
| 22:10.53 | [TK]D-Fender | NightKhaos: How so? I jsut told you exactly what kind of problems you might expect from that card. |
| 22:11.42 | [netman] | my telco says it's allowing me change CID, but I cannot... what could be the issue? |
| 22:12.18 | *** part/#asterisk gbr_ (n=gbr@200.103.96.98) |
| 22:12.50 | NightKhaos | [TK]D-Fender: with all do respect you only told me after I told you my father had gone ahead and ordered one. |
| 22:13.04 | j0 | [netman]: hey.. same problem :) |
| 22:13.52 | [netman] | pri, isn't it? :) |
| 22:13.52 | NightKhaos | [TK]D-Fender: I am also getting some contrary advice from other people saying the X100P is fine. So I want to see some facts, and google isn't be helpful today. |
| 22:13.56 | [TK]D-Fender | NightKhaos: So now you're forwarned. Others apparantly warned you as well. You seem to good ignoring things you don't want to hear |
| 22:13.59 | j0 | [netman]: yup |
| 22:14.28 | [netman] | I tried several methods of changing caller id... without any results |
| 22:14.30 | [TK]D-Fender | NightKhaos: the X100P's are VERY hit-or-miss. |
| 22:14.38 | Qwell | mostly miss |
| 22:14.43 | [netman] | I'm absolutely sure they r not allowing me to change it |
| 22:14.43 | j0 | NightKhaos: add me to the list of people who don't reccomend x100p's.. like netman said. test and play only |
| 22:15.07 | j0 | [netman]: what pri card are you using? |
| 22:15.31 | Qwell | [netman]: what are you trying to change, exactly? Just number? |
| 22:16.22 | [netman] | TE405P |
| 22:16.28 | NightKhaos | [TK]D-Fender: I do hear you now. I'm just pissed off is all, okay? Thing is already payed for, can't cancel the order... and now I'm getting told there is a high chance it'll be a dud. Wouldn't you be upset? |
| 22:16.52 | [netman] | Qwell: I'm trying to change my outdound CID |
| 22:16.57 | [netman] | Qwell: I'm trying to change my outbound CID |
| 22:16.58 | Qwell | just number? |
| 22:17.16 | [TK]D-Fender | NightKhaos: Your self-control issue, not mine. Sorry for your loss. Suck it up and hope for the best while preparing for the worst. |
| 22:17.26 | j0 | [netman]: sangoma A101d here |
| 22:17.26 | [netman] | the DID asigned to my PRI |
| 22:18.00 | Qwell | [netman]: trying to set it to a number in the block assigned to that PRI? |
| 22:18.09 | Qwell | often, they won't let you just set it to something arbitrary |
| 22:18.28 | [netman] | Qwell: I did it before the telco recognise the had to allow me |
| 22:18.42 | [netman] | not really after they "completed" his work |
| 22:18.54 | [netman] | but I tell them I need to use special numbers |
| 22:19.10 | [netman] | so , if the did it, I should use *any* CID |
| 22:21.10 | *** join/#asterisk UnixDog (n=UnixDog@181.128.204.68.cfl.res.rr.com) |
| 22:21.56 | [netman] | Qwell: I have just tried now |
| 22:22.11 | j0 | [netman]: can you set it to your assigned DID's? |
| 22:22.13 | [netman] | They allow me to use any DID in the PRI range as CID |
| 22:22.20 | [netman] | yes j0 |
| 22:22.32 | Qwell | then it's a problem on their end |
| 22:22.32 | [netman] | but that isn't what I tell them |
| 22:22.36 | j0 | aah.. i can't set it to anything :) |
| 22:22.42 | j0 | it just uses the "main" line |
| 22:22.51 | [netman] | but, it is a telco trouble, isn't it? |
| 22:23.01 | j0 | [netman]: that's what everyone is saying |
| 22:23.02 | NightKhaos | [TK]D-Fender: when you say the X100P, do you mean Digdium clones? |
| 22:23.03 | [TK]D-Fender | j0: And the reason you aren't showing us exactly what you're doing at CLI with PRI debug is.....? |
| 22:23.04 | [netman] | or is it mine? |
| 22:23.21 | [TK]D-Fender | NightKhaos: ALL of the X100 family, "clone" or otherwise. |
| 22:23.23 | j0 | [TK]D-Fender :) sec |
| 22:23.28 | [TK]D-Fender | SMRT |
| 22:23.29 | [netman] | j0: but , is it possible to set to anything or not? |
| 22:23.43 | j0 | [netman]: only what your telco allows |
| 22:23.46 | [TK]D-Fender | People shoud stop assuimg its his telco's fault when I don't trust his CODE :p |
| 22:23.53 | [TK]D-Fender | should* |
| 22:23.58 | [netman] | j0: but my telco said they allows me |
| 22:24.07 | [netman] | but I don't see it's true |
| 22:24.09 | [TK]D-Fender | Geez people...ask him for OUTPUT <- |
| 22:24.15 | *** join/#asterisk obnauticus (n=obnautic@c-67-160-181-107.hsd1.wa.comcast.net) |
| 22:24.27 | [TK]D-Fender | swears everyone has lost their debugging skills |
| 22:27.48 | [netman] | j0: but do u know anybody who can set any CID in his PRI? |
| 22:28.40 | j0 | [netman]: no |
| 22:28.55 | [netman] | me neither |
| 22:29.05 | j0 | I see where it sets the caller id (6045571489 on line 21) but it doesn't show up on the telco. http://pastebin.com/m4e2ae2f5 |
| 22:30.48 | j0 | what could this mean on line 42: < Ext: 1 Cause: Access information discarded (43), class = Network Congestion (resource unavailable) (2) ] |
| 22:31.44 | *** join/#asterisk RoyK (n=roy@ti211210a080-3097.bb.online.no) |
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| 22:33.56 | [netman] | what sort of debug have u enabled to log that lines j0 ? |
| 22:35.11 | j0 | [netman]: pri debug span 1 |
| 22:35.17 | [netman] | thx |
| 22:35.18 | j0 | type that on the console |
| 22:35.36 | [netman] | I'm gonna try |
| 22:36.42 | edwin_quijada | anybody has installed swift to TTS? |
| 22:37.41 | [netman] | I got |
| 22:37.43 | [netman] | Presentation: Presentation allowed of network provided number (3) '902902902' ] |
| 22:37.58 | [netman] | but I really don't see that number |
| 22:38.21 | j0 | i wonder if that's exactly the same thing mine is saying.. i get Presentation: Presentation permitted, user number not screened (0) '6045571489' ] |
| 22:39.04 | [netman] | but I also watch : |
| 22:39.04 | [netman] | <PROTECTED> |
| 22:39.23 | j0 | it's all greek to me |
| 22:40.06 | [netman] | but with the same message |
| 22:40.07 | [netman] | <PROTECTED> |
| 22:40.25 | [netman] | I can set a CID if it's one of my DIDs |
| 22:41.52 | RoyK | [netman]: seems you should call the telco |
| 22:42.10 | [netman] | RoyK: the telco says they allowed me to set the CID |
| 22:42.13 | [netman] | twice |
| 22:42.13 | *** join/#asterisk whye (n=whye@unaffiliated/whye) |
| 22:42.26 | [netman] | friday and today monday |
| 22:42.28 | RoyK | [netman]: they have told me that a few times as well even when it didn't work |
| 22:43.15 | RoyK | then I sent them a dump and it took an hour and it was fixed :P |
| 22:43.22 | [netman] | can I ask them for showing me it really works? |
| 22:43.45 | RoyK | just send them pri debug output |
| 22:43.48 | RoyK | that usually helps |
| 22:44.06 | [netman] | but my pri debug says |
| 22:44.10 | [netman] | <PROTECTED> |
| 22:44.23 | [netman] | but I don't see that number on my phone |
| 22:45.21 | [netman] | so the debug say what they want to say, not the real thing |
| 22:46.01 | RoyK | the debug shows the asn.1 codes sent over the wire |
| 22:46.05 | *** join/#asterisk DarkRift (n=dark@bas10-montreal02-1177583238.dsl.bell.ca) |
| 22:46.19 | [netman] | OK RoyK |
| 22:46.30 | [netman] | so I guess It really could help me |
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| 22:47.16 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 22:47.44 | RoyK | the numbers in front of the text is the raw ans.1 data - the text is the decoded stuff |
| 22:50.26 | j0 | RoyK: ah :) thanks for that tip |
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| 22:54.35 | [netman] | thx RoyK |
| 22:55.10 | [netman] | good night all |
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| 23:00.16 | s0lid | is there a way in asterisk to send cause code 34 in SIP when the line didn't got through? |
| 23:00.26 | s0lid | my switch always recieve cause code 21 |
| 23:00.45 | s0lid | i tried to set the DIALSTATUS=CONGESTION but still i get cause code 21 |
| 23:01.15 | *** part/#asterisk Xamusk (n=Xamusk@189.1.136.223) |
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| 23:08.28 | eXistenZ | [TK]D-Fender, hey |
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| 23:53.09 | The_TiK | is there a variable that containts the number I am calling? |
| 23:53.41 | The_TiK | CALLERID(num) contains the number i am calling from but im tryin to set the filename for a number I am calling to |
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| 23:56.57 | lmadsen | The_TiK: the answer is probably ${EXTEN{ |
| 23:57.01 | lmadsen | errr... ${EXTEN} |
| 23:57.43 | The_TiK | i thought that would be the exten that I am calling from |