00:01.53 | coolhp | Is anyone having problems getting the blindxfer feature to work ? I have "Tt" passed as options in my Dial() command and DTMF is set to RFC2833 on both sides... automon works if I set DYNAMIC_FEATURES but blindxfer never does :-( |
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00:14.02 | LemensTS | . |
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00:20.05 | actros1840 | hi |
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00:22.16 | Hyphenex | What is AMP and would I have it if I built asterisk from source? |
00:22.41 | ManxPower | ~amp |
00:22.41 | jbot | amp is, like, NOT supported here! People using it should join #freepbx (FreePBX is the new name of AMP) |
00:23.29 | ManxPower | coolhp: try "Ww |
00:27.33 | Hyphenex | okies, any guides on getting asterisk to work with FWD without AMP then? |
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00:37.26 | ManxPower | AT&T sent me an e-mail advertizing a new product. It was so filled with crappy HTML that my html cleaning script basically deleted the entire contents of the message. |
00:37.45 | ManxPower | Hyphenex: Um, have you tried the FWD site or the Wiki? |
00:38.07 | ManxPower | All AMP is, is a GUI for pussies that don't want to edit config files. |
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00:38.41 | kiscokid | I have a Polycom phone question |
00:39.40 | ManxPower | kiscokid: ask it |
00:39.41 | kiscokid | Anyone know what makes the call timer start on a Polycom? Sometimes it doesn't start right away. |
00:40.04 | ManxPower | kiscokid: no idea, I would imagine it would be trigered by the far end answering |
00:40.11 | kiscokid | right |
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00:40.39 | kiscokid | but sometimes when I call a 800 number the signal is not received for a long time |
00:40.43 | actros1840 | nobody use MP3Player ? |
00:41.16 | sbingner | kiscokid, I've called 1800 numbers where they don't actually answer till I choose the first option |
00:41.24 | kiscokid | seems like if the signal is not sent by the remote end within 60 secs the phone decides to hang up |
00:41.26 | sbingner | I think they're cheating on long distance charges that way |
00:41.44 | Hyphenex | Little question. Can a user belong to more then one contexts? |
00:42.32 | kiscokid | Does the Polycom or * have some parameter that I could change to make it wait for 120 secs? |
00:44.01 | kiscokid | also, any know the name for that signal? |
00:44.59 | kiscokid | sbinger: the particular 800 number I am having trouble with doesn't send the signal until way into the call menu |
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01:05.18 | stansmith | 0 |
01:06.22 | ManxPower | kiscokid: put an Answer before the Dial |
01:08.33 | ManxPower | kiscokid: I suggest you read thru the Admin Guide to see if anything looks useful. |
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01:14.29 | ManxPower | kiscokid: You are NOT the only one that has calls that ring for more than 60 seconds |
01:16.06 | stansmith | lol |
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01:25.06 | kamitodo | Just got a second-hand SPA2102. My setup is DSL modem -> OpenWrt -> SPA2102 plus all other LAN and WIFI devices. Cannot access the SPA2102 via the browser. Any ideas? |
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01:26.52 | lyroy | Does someone here have experience on how to setup a provisioning server for linksys phone adapter like pap2 and wrt54gp2 ? |
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01:33.58 | kiscokid | ManxPower: ? |
01:35.20 | kiscokid | ManxPower: should I put Answer before Dial for every call? |
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01:38.57 | stansmith | lol? |
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01:49.00 | ManxPower | kiscokid: that is up to you. |
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01:55.24 | kamitodo | Is anyone familiar with SPA2102? |
01:56.23 | kiscokid | Manxpower: I don't understand the implications of your answers. |
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02:02.27 | MatBoy | it's really weird that when I use in a2billing my cardnumber and my password, I never can authenticate |
02:02.34 | MatBoy | always password wrong in the errorlog |
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02:10.46 | mchou | anyone here have experience with linksys wrtp54g (wireless router with 2 voice ports)? |
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02:11.36 | coppice | I have one |
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02:12.12 | mchou | coppice: what ver firmware you have loaded on it? |
02:12.27 | MatBoy | mhh, I don't get it, maybe I need sleep :) |
02:12.39 | coppice | something old. they never seem to update it. it sucks |
02:12.39 | MatBoy | but this is actually quite easy to setup with a2billing |
02:13.01 | mchou | coppice: you need frequent resets for that device? |
02:13.27 | mchou | coppice: like it "crashes" on you? |
02:13.40 | MatBoy | when you can login to a2billing as a user, can you also authenticate with the same details ? |
02:13.53 | coppice | yeah. everyone complains about the same thing. if you never make a call it seems to work Ok as a wireless router, it seems to be telephony that makes it fall over |
02:14.14 | mchou | coppice: sigh. |
02:14.43 | mchou | coppice: is pap2 more relaible or just as bad? |
02:14.52 | mchou | reliable* |
02:15.07 | coppice | i don't hear complaints about the PAP2 crashing |
02:15.41 | mchou | coppice: crap. maybe I should have bought pap2 instead. |
02:15.51 | MatBoy | yeah !! |
02:15.53 | MatBoy | did it :) |
02:16.17 | MatBoy | guys I love you :P |
02:16.38 | TJNII | mchou: I have a couple PAP2s. No complaints. All eBay specials. |
02:17.24 | mchou | TJNII: what's the going price for those babies? |
02:18.12 | TJNII | Depends on demand. The last ones I got were $30ea, but I paid too much. |
02:18.21 | mchou | and is the dlink VTA the same thing? cause if so I'd like to just go to Fry's to pick them up |
02:18.47 | coppice | mchou: I'd advise against getting babies. they are even more troublesome than those linksys boxes |
02:19.09 | TJNII | Don't know. You just have to watch for firmware locks when you bou off the shelf. If you're willing to hack, more power to you. |
02:19.49 | mchou | well, apparently the dlink vtas are harder to hack for whatever reason |
02:20.14 | mchou | event though some have said the HW is exactly the same |
02:20.27 | mchou | as a pap2, that is |
02:21.51 | mchou | voip HW is costing me a mint, testing junk that's not reliable :( |
02:22.19 | mchou | had such high expectations for wrtp54g |
02:23.29 | TJNII | Heh. I've had good luck with the pap2s. I've got a bunch of budgetone-100s that are good if you overlook the poor caller ID and speakerphone. |
02:24.02 | TJNII | Some buddies of mine have some nice polycom-500s, but they manage to find a mislabled eBay auction to get them cheap. |
02:24.43 | mchou | I dont think I'm ready for a full blown IP phone yet at this stage |
02:24.55 | mchou | still prefer ATAs |
02:25.16 | TJNII | I got a SNOM 220 yesterday, I have high hopes for that. |
02:25.17 | mchou | just cause voip HW is such a crapshoot |
02:25.44 | TJNII | Heh. Doing an ebay search for sip phone hasn't led me wrong yet. |
02:25.50 | mchou | TJNII: new or used? |
02:26.03 | TJNII | Well, except that IP0021 phone. That thing is junk. |
02:26.24 | coppice | all current VoIP hardware is crappy. its just a question of how crappy. one or two phones are starting to look like their maker has a clue, though |
02:26.38 | TJNII | New |
02:27.20 | TJNII | The SNOM core is just a powerPC chip running linux. I probably won't be able to resist hacking it. |
02:27.47 | mchou | coppice: which phone makers have a clue? |
02:27.59 | coppice | the power PC seems to be growing in these little embedded boxes |
02:28.42 | coppice | well, in some markets polycom seems to sell decent phone for <$100. most people say nice things about the value for money of those |
02:29.18 | TJNII | The Polycom 500s look nice, usually go for around $100 on eBay |
02:29.27 | _ShrikE | IMHO polycom's low end phones have been crap until the 320/330's came out. |
02:29.54 | coppice | yeah, those 320s seem to be designed to shake up the market |
02:30.49 | TJNII | I say the budgetones make good starter phones. Not expensive, good call quality on the handset, easy to set up. |
02:31.04 | TJNII | Though I know "budgetone" is a naughty word here....... |
02:31.08 | kiscokid | Grandstream? |
02:31.11 | TJNII | yea |
02:31.16 | kamitodo | anyone familiar with SPA2102? I got version firmware version 5.1.12 and don't know the default admin pass. |
02:31.21 | mchou | TJNII: so what exactly is wrong with caller ID on budgetone? |
02:31.27 | TJNII | Numeric only |
02:31.35 | TJNII | No letters, just the number |
02:31.43 | coppice | I think if grandstream just got a competant designer to do a nicer case for them, people would have a far higher opinion of them |
02:31.53 | TJNII | Heh |
02:32.14 | TJNII | Mine look kinda like hotel phones. They look fine in the bedroom, not so much on my desk. |
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02:33.34 | mchou | kamitodo: is it a locked version? |
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02:34.45 | kamitodo | mchou: I don't know. Bought a used one off amazon. How do I tell? |
02:34.53 | mchou | lol |
02:35.02 | mchou | if it's used it's locked :) |
02:35.29 | mchou | kamitodo: you better read up on how to hack it |
02:35.30 | kamitodo | well, is it for sure? how do i check? |
02:35.53 | kamitodo | hm... any experience with that? |
02:36.14 | mchou | kamitodo: plsa do your own homework on that |
02:36.18 | mchou | pls* |
02:36.57 | mchou | plenty of sites give general ways to unlock the locked devices. |
02:37.00 | mchou | YMMV |
02:37.35 | kamitodo | well, OK. what if it is not locked? what's the default pass then? |
02:39.12 | mchou | kamitodo: should be in the linksys documentation |
02:39.39 | mchou | try admin/admin |
02:42.20 | kamitodo | admin/admin does't work. |
02:42.37 | kamitodo | i see on the box hat the model no. is spa2102-na |
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03:08.22 | TJNII | GAAH! |
03:08.41 | TJNII | Never was good with 'dem things. |
03:09.03 | obnauticus | IS anyone here good with a2billing? |
03:09.32 | kamitodo | there *must* be a way to tell whether the damn thing is locked! referring to the spa2102 of course. |
03:09.36 | obnauticus | because for some reason when i add a prefix as a rate, a2billing is still not forwarding the call. |
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03:15.15 | WilliamK | kamitodo, tried the manual for it yet? |
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04:07.12 | outtolunc | .. |
04:07.24 | drmessano | .. |
04:07.54 | drmessano | @? |
04:08.07 | outtolunc | as to the lack of scrolling *not* happening here |
04:08.52 | drmessano | ah |
04:09.07 | drmessano | Things have been quieter |
04:09.53 | coppice | ......zzzzzzZZZZZZZZ |
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04:25.37 | Loof | Anyone here familiar with comdial PBX systems... it seems that I can put a T1 interface card on an asterisk box and basically have talk T1 interface to T1 interface as a sort of pass-through |
04:25.52 | Loof | At least, that's how I'm understanding the description I'm seeing |
04:25.57 | Loof | am I offbase here? |
04:27.38 | Loof | hmm |
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04:48.16 | x86 | what's comdial? |
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04:58.59 | rfernandez | hi! a p 4 with 3.0 ghz and 1gb in ram may handle 24 trunks with 24 extensions? |
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05:28.44 | obnauticus | how do i pass inline operators for agi scripts from the dialplan? |
05:29.04 | obnauticus | like say i have script.agi then ./script.agi x y z |
05:30.19 | obnauticus | and i want to pipe stdout from that script into Festival, so how do I do that? |
05:31.36 | obnauticus | wow |
05:32.04 | obnauticus | that crapped on my question |
05:32.04 | outtolunc | bummer |
05:32.17 | obnauticus | do you know |
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05:34.59 | outtolunc | i'd have to scroll back |
05:35.24 | outtolunc | use | |
05:36.02 | outtolunc | exten => _X.,5,AGI(dnidlookup.agi|${CALLERID(num)},${IF(${ISNULL(${CALLERID(dnid)})}?${EXTEN}:${CALLERID(dnid)})}) |
05:36.06 | outtolunc | as an example |
05:37.09 | outtolunc | another example |
05:37.17 | outtolunc | exten => 7,n,AGI(disporec.agi|${OPTION},"399",${CAMPAIGN},${LEADID},"10") |
05:37.27 | obnauticus | outtolunc like say i want a user to be able to enter a number |
05:37.30 | obnauticus | into their keypad |
05:37.33 | obnauticus | i do a waitexten |
05:37.41 | obnauticus | and you use that number as an in-line operator |
05:37.43 | obnauticus | how do i do that |
05:37.46 | obnauticus | or an argument |
05:37.56 | outtolunc | you have to use read for that |
05:38.20 | outtolunc | or background |
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05:38.57 | obnauticus | then how do i pipe the stdout from the script |
05:38.57 | obnauticus | to agi |
05:38.57 | obnauticus | then how do i pipe the stdout from the script to agi |
05:39.00 | outtolunc | you don't you read it to a var, then use the ${var} when you get back into dialplan |
05:39.47 | obnauticus | >: | |
05:39.47 | obnauticus | what do you mean? |
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05:40.26 | outtolunc | http://dynx.net/ASTERISK/gnudialer/agiIVR.agi |
05:40.26 | outtolunc | read that |
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05:40.28 | obnauticus | k |
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05:40.28 | outtolunc | it give *alot* of examples |
05:40.28 | outtolunc | gives even |
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05:41.04 | outtolunc | search for 'question_1' that directly relates to your question |
05:41.04 | obnauticus | thanks |
05:41.31 | obnauticus | so like |
05:41.37 | obnauticus | the read sets the DTMF tones to variables |
05:41.44 | obnauticus | read from the user's channel? |
05:42.04 | outtolunc | yes |
05:42.10 | outtolunc | play a prompt |
05:42.20 | outtolunc | *read* input |
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05:42.27 | outtolunc | store to var |
05:42.29 | obnauticus | http://www.voip-info.org/wiki/view/Asterisk+cmd+Read |
05:42.32 | obnauticus | im reading that right now |
05:42.35 | obnauticus | and your other script |
05:44.44 | obnauticus | outtolunc so how do i store stdout to a variable? |
05:45.38 | outtolunc | you mean stdin |
05:45.45 | obnauticus | no stdout from the agi script |
05:45.50 | outtolunc | you don't |
05:45.53 | obnauticus | :\ |
05:45.58 | obnauticus | I have an NPA-NXX lookup thing im trying to make.. |
05:46.07 | outtolunc | you store stdin, to the var as use it as ${VAR} |
05:47.49 | outtolunc | http://dynx.net/ASTERISK/AGI/agi-ccard.agi |
05:48.05 | obnauticus | http://www.monetra.com/~brad/callerid_shell.agi |
05:48.10 | obnauticus | im actually trying to get that to work |
05:48.11 | obnauticus | :\ |
05:48.26 | *** join/#asterisk horseradish (i=user@63-76-119-176.directcom.com) |
05:49.19 | horseradish | i need advice on an asterisk setup w/ equipment |
05:49.45 | outtolunc | look at the last line |
05:50.19 | outtolunc | once that agi returns to the dialplan you use it as ${lookupname} |
05:50.39 | obnauticus | well see i want the user to input their own number. |
05:50.43 | obnauticus | because right now it just gets it from the callerid |
05:50.56 | obnauticus | caller id number* |
05:51.03 | obnauticus | i.e. ; exten => s,1,AGI(callerid_shell.agi|${CALLERIDNUM}) |
05:51.05 | horseradish | i want a 2 line SOHO setup. with 2 voicemail boxes, an introductory message when ppl call, ability to place people on hold, send them to voicemail boxes, transfer calls, and an after-hours message |
05:51.05 | outtolunc | yes, you supply a NUMBER, it finds a NAME |
05:51.18 | obnauticus | ya im making my * server to read for a number |
05:51.30 | horseradish | i'm not sure if it would be better to buy my own box and run it on a POTS line or if i should go with a VoIP provider |
05:52.25 | outtolunc | horseradish: note that using the available hardware on standard 1mb's *usually* does not give you things like disconnect supervision |
05:53.14 | obnauticus | outtolunc oh i know what you mean |
05:53.26 | outtolunc | bingo! |
05:53.27 | obnauticus | it then lets you use it later as $lookupname |
05:53.28 | outtolunc | hehe |
05:53.32 | obnauticus | err |
05:53.33 | obnauticus | cool |
05:53.38 | outtolunc | as ${lookupname} |
05:53.43 | obnauticus | and that would be |
05:53.57 | outtolunc | exten => s,1,AGI(callerid_shell.agi|${CALLERIDNUM}) |
05:54.02 | obnauticus | exten => s,1,Festival(${lookupname}) |
05:54.16 | obnauticus | http://pastebin.ca/924299 |
05:54.16 | obnauticus | that |
05:54.18 | obnauticus | s what i haev written |
05:54.19 | outtolunc | exten => s,2,NoOp(callerid name is ${lookupname}) |
05:54.45 | outtolunc | you can't have 2 s,1's |
05:54.47 | obnauticus | and the script can read the number as NPA-NXX-XXXX |
05:54.56 | obnauticus | i know that i was going to edit the priorities later |
05:54.57 | obnauticus | lol |
05:55.15 | outtolunc | you got 3 |
05:55.25 | obnauticus | because i knew it was going to be priority 3 |
05:55.36 | obnauticus | i write it, then i do priorities... it's my thing |
05:55.37 | obnauticus | lol |
05:55.49 | outtolunc | extension processing on a 'reload' stops at the first error |
05:56.01 | outtolunc | (which you should see with verbose 3 or higher on the cli) |
05:56.03 | obnauticus | i would have noticed |
05:56.08 | obnauticus | ya |
05:56.14 | obnauticus | im just an AGI nub |
05:56.14 | obnauticus | lol |
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05:56.38 | outtolunc | just use s,1 then the rest use s,n |
05:56.51 | obnauticus | heh |
05:57.50 | obnauticus | errrr |
06:00.47 | obnauticus | outtolunc how do i make it so it also echo's the Company name |
06:01.22 | obnauticus | wait i think it already does for ${company} |
06:01.41 | outtolunc | if the script returns that also, store it to a variable also (look at the last line) |
06:02.39 | obnauticus | i don't think it returns that |
06:02.39 | obnauticus | :"\ |
06:03.34 | outtolunc | it doesn't 'myname' gets replaced along the way with whatever 'name ish' thing it finds |
06:04.42 | obnauticus | So :\ |
06:06.14 | obnauticus | oh outtolunc you can change the "lookup order" |
06:06.18 | obnauticus | so it finds 411 info too |
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06:06.36 | outtolunc | yes, i get that <G> |
06:07.17 | horseradish | is vonage a good voip provider for asterisk stuff |
06:07.30 | obnauticus | no, use voipjet |
06:07.33 | horseradish | or what is a good voip provider for my 2 line soho |
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06:11.09 | Loof | So, can a T1 interface talk to another? I.e. something that expects to hook up to the c/o T1 talking to a T1 interface on an Asterisk box? |
06:11.28 | jql | Loof: yes |
06:12.05 | horseradish | if i pay for a voip provider, then asterisk providers all the features like call waiting, call queing, hold music, etc. right? |
06:12.09 | jql | ideally, you're using pri, which means you set the asterisk box to transmit pri-net |
06:12.15 | horseradish | and i can have unlimited lines and phone numbers if i go voip, right? |
06:12.34 | horseradish | can i get a 1-800 number with someone like voipjet |
06:12.44 | scooby2 | anyway to flap or loop a t1 on a sangoma card? |
06:12.45 | Loof | Ok, that makes sense of what I'm seeing in the FAQs then |
06:13.42 | Loof | So if we've got a dual T1 PBX and want to keep our existing system and use DID for call routing... we need a quad T1 card on the asterisk box |
06:13.58 | Loof | 2 in from c/o and 2 out to the old PBX |
06:14.35 | TJNII | horseradish: Well, not unlimited. There are bandwidth restrictions and restriction from your sip provider. |
06:14.41 | TJNII | And yes you cen get a 1800 |
06:14.46 | jql | Loof: that would make sense, yes |
06:15.21 | jql | Loof: I'm sure the FAQs and manuals all mention it, but make sure your quad card gets its clock from the c/o |
06:15.31 | TJNII | hmmm.... fwd and iax just doesn't wanna go...... |
06:15.43 | Loof | I've been trying to find a good FAQ or manual or howto for it |
06:16.02 | Loof | I've found forum answers that -seemed- to indicate that |
06:16.06 | Loof | but I wasn't sure if that work :P :) |
06:16.15 | jql | I'll confirm it. :) |
06:16.28 | Loof | Do we need a csu/dsu between the PBX and the asterisk box? |
06:16.38 | Loof | I'd assume not, from what I see of the ports. |
06:17.03 | Loof | And I would guess we need a straight wire cable, not a cross over |
06:17.09 | jql | no, just needs cat5 cable |
06:17.32 | jql | T1 cross-over is different than ethernet crossover |
06:17.35 | jql | and you do need it |
06:17.42 | jql | get a wiring diagram |
06:19.08 | Loof | Duh, found it :) |
06:19.09 | drmessano | IIRC you swap orange and blue |
06:19.22 | Loof | Should've googled for t1 crossover in the 1st place :) |
06:19.26 | jql | heh |
06:19.55 | Loof | The only thing I'm not sure of... what happens when someone hits a transfer button or vm button within the pbx system |
06:20.13 | Loof | Does that pass back up the chain in some way? |
06:20.26 | jql | not likely |
06:20.33 | Loof | i.e. will we be able to assign extensions to and transfer between voip phones and the pbx phones? |
06:20.37 | jql | the phone company doesn't care about your voicemail |
06:21.11 | jql | your voip phones can certainly dial the pbx with enough motivation |
06:21.47 | jql | the pbx may require voodoo to know to dial out for what look like internal extensions |
06:21.58 | jql | they all have the voodoo, but it's always voodoo |
06:22.08 | jql | prepare a virgin for sacrifice |
06:22.31 | Loof | Ok, so from the perspective of asterisk these look like a whole bunch of POTS interface phones... and any voip phones are what they are. |
06:22.57 | jql | pretty much |
06:23.28 | Loof | Thus we'd have some kind of DTMF interface to the functions... #<something>,<ext> to transfer while on a call or something |
06:23.47 | jql | quite practical of you |
06:23.56 | Loof | Sorry, not a telecom guy :) |
06:23.59 | Loof | If its not %100 obvious |
06:24.09 | Loof | I'm on the IT side of the universe with a little telecom experience |
06:24.17 | Loof | enough to hurt my head provisioning a T1 now and then |
06:25.03 | jql | t1 here, and a pbx there, and soon you're talking business |
06:25.04 | Loof | And I think virgin sacrifice is against company policy... we can request chicken entrails with forms in triplicate |
06:25.07 | Loof | ;) |
06:25.42 | jql | entrails entreat the spirits, but virgins appease the Gods |
06:25.43 | Loof | Ok, so from the perspective of the end users... if we do this right... they will end up with a few dozen 'useless' buttons on their phones. |
06:26.07 | Loof | Something that will actually work transfering pbx<->pbx... but fail pbx<->sip device. |
06:26.22 | Loof | Thus, we'll retrain to use the DTMF codes for transfer, vm access, etc. |
06:26.36 | jql | depends on how opaque the pbx configuration is |
06:26.59 | jql | sometimes they have a mapping of extension 4xxx => dial 7005554xxx |
06:27.01 | Loof | Unless the phones can be reprogrammed to do some sort of signaling via DTMF for the call features |
06:27.06 | jql | which would go to asterisk |
06:27.07 | Loof | but that seems like too much to ask for |
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06:28.43 | Loof | it's a comdial DXP |
06:29.16 | Loof | So, I think that type of mapping should work... but call transfer and VM are internal, as far as I gather |
06:32.46 | jql | old school pbxs are old school |
06:33.44 | jql | that comdial has an 1100 page manual |
06:34.59 | Loof | I wonder if it has some way to do in-band signaling then... |
06:35.00 | Loof | hmm |
06:35.41 | Loof | I assume there is no other way to salvage the phone and wiring infrastructure without continuing to use the PBX controller |
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06:37.09 | jql | if there is, the PBX installer was a fool |
06:37.16 | jql | err... I mean... perhaps. :) |
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06:38.28 | jql | well hell |
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06:38.42 | jql | you can indeed reprogram all the buttons on the phone using this disgusting manual. :) |
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06:48.08 | Star568 | please teach me how to autostart asterisk on centos version 4 |
06:49.25 | BeeBuu | Star568: run asterisk |
06:49.42 | *** part/#asterisk horseradish (i=user@63-76-119-176.directcom.com) |
06:51.28 | Star568 | i want asterisk can auto start the service after system reboot |
06:54.30 | BeeBuu | add it in /etc/init.d/ |
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07:03.53 | Star568 | BeeBuu: i am a newbie to linux, where and how can i find the startup script for asterisk, just cp the file to this directory? |
07:04.12 | _mwoodj_ | Star568: 'make config' will install the scripts. 'chkconfig asterisk on' will enable it for centos |
07:05.49 | _mwoodj_ | See the "Installing Asterisk" section of this document: http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation |
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07:06.48 | Star568 | thanks all for the information :) |
07:06.53 | _mwoodj_ | no problem |
07:17.03 | Loof | jql: Thanks for the help, that was exactly the boost I needed to figure out the rest. |
07:18.44 | Star568 | _mwoodj_: it works now :D |
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09:05.41 | BeeBuu | ~book |
09:05.41 | 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 |
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09:31.12 | BeeBuu | anyone still here now? |
09:31.20 | obnauticus | /me |
09:31.44 | BeeBuu | can i dial on console? |
09:31.51 | obnauticus | via console? |
09:31.55 | BeeBuu | ya |
09:32.05 | obnauticus | you can use err |
09:32.08 | obnauticus | Originate. |
09:32.17 | obnauticus | origionate with the application dial |
09:32.43 | BeeBuu | can i dial with Priority under console? |
09:33.40 | BeeBuu | here is what i want: make the * system auto call some people and play a message |
09:34.08 | BeeBuu | or meet them |
09:34.09 | obnauticus | you can make an agi script to run in the background |
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09:34.26 | BeeBuu | a morning meet... :-) |
09:34.42 | obnauticus | that osunds wrong |
09:34.55 | BeeBuu | obnauticus: what's wrong? |
09:35.02 | obnauticus | erg |
09:35.14 | BeeBuu | erg? |
09:35.21 | obnauticus | erg. |
09:35.52 | BeeBuu | what's wrong? |
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11:25.58 | _gm | anyone here tries realtime ldap driver? |
11:26.01 | _gm | tried* |
11:39.56 | ^shark_ | hi guys i am trying to use the cfgfmt.exe tool to change the .txt file to binary for the 7912g phone but i dont understand what -FLAG mean in here >> cfgfmt.exe -FLAG sip_ptag.dat gkdefault.cfg.txt gkdefault.cfg, is it an option of its self? i certainly dont see it reflected here >> http://cisco.com/en/US/docs/voice_ip_comm/cuipph/7905g_7912g/3_3/h323/english/administration/guide/7905conf.html#wp1052210 |
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11:50.49 | ^shark_ | is this (-FLAG) supposed to mean an RC4Password ? or something else? |
11:52.38 | ^shark_ | or nothing at all >> does this make more sense if i am not using RC4Password -- cfgfmt.exe -sip_ptag.dat gkdefault.cfg.txt gkdefault.cfg |
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12:02.00 | ^shark_ | i am getting error >> error: can't open ptag file: 'ptag.dat' who can kindly help me with this? |
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12:06.19 | ^shark_ | any body atleast read my query? |
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12:36.15 | ice_croft | !book |
12:36.23 | ice_croft | ~book |
12:36.24 | 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 |
12:37.16 | ice_croft | hi all |
12:39.59 | ice_croft | where can i read some about * interconnection? |
12:40.09 | ice_croft | *'s |
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13:03.45 | ecto | ice_croft: what do you mean? IAX connections, SIP connections, etc? |
13:04.00 | ecto | Setting up an IAX connection between two Asterisk boxes is in TFOT book |
13:04.04 | riddlebox | can anyone help me with sla? I have configured it according to doc/sla.pdf and when I do sla show trunks and sla show stations there is nothing, http://pastebin.ca/924527, there is nothing |
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13:15.44 | ice_croft | ecto> i found it already, thanx |
13:23.20 | ^shark_ | hi |
13:24.42 | ^shark_ | i am trying to boot the 7912g ip phone, and i have my dhcp/tftp server (192.168.100.166) and the files that the phone is looking for which you can see from this >> http://internetworkpro.org/pastebin/2013 |
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13:25.05 | ^shark_ | but still the phone will not boot, any ideas why this is happening? |
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14:04.53 | *** join/#asterisk dacs (n=haiger@unaffiliated/dacs) |
14:05.09 | dacs | good morning every one |
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14:08.19 | dacs | i have been asked by our church priest, to find a telephone device that can store 100+ phone numbers so that he can record a message regarding service time and date and send it at once to all of the numbers.! is that possible to do with *? |
14:10.36 | _ShrikE | dacs: yes, it's possible |
14:11.11 | riddlebox | dacs, astercrm |
14:12.11 | dacs | riddlebox: where can i get more info please |
14:12.23 | riddlebox | www.astercrm.org |
14:12.53 | riddlebox | dacs, the developer is working on getting it to work with fxo trunks now, may be a couple of weeks |
14:12.59 | riddlebox | but it works with sip trunks now |
14:13.01 | dacs | also what is confusing me , how is that possible :) , shouldn't the phone ring first to go to vmail |
14:13.36 | riddlebox | so the priest wants to call vmail boxes? or call peoples houses? |
14:14.13 | dacs | riddlebox: people houses /cell . because right now he dails every single phone, |
14:14.32 | riddlebox | asterisk+astercrm, will be able to handle it |
14:15.09 | dacs | riddlebox: or maybe be like IVR , call all those numbers at once and play the msg to who ever picks |
14:15.09 | _ShrikE | dacs: If he wants to send the 100 calls all at once, you will need to get him a voip provider or about 4 T1s |
14:15.11 | riddlebox | when solo, gets the fxo trunks to do it |
14:15.26 | riddlebox | you can set it and forget it! |
14:16.16 | riddlebox | dacs, my boss has me creating an asterisk box with 4 fxo ports in order to that very thing |
14:16.17 | dacs | riddlebox: i like the solo, since expness will come out of my poket, i hate it when priest do that |
14:16.49 | riddlebox | well how much are yo willing to spend? |
14:16.54 | dacs | they tell you we need this , this and that and GOD bless you |
14:17.00 | riddlebox | hey _ShrikE have you ever dealt with sla? |
14:17.46 | dacs | riddlebox: there is an old box in the church i will check it and see if i can load linux and then * |
14:17.47 | _ShrikE | I try to avoid it :) |
14:18.26 | riddlebox | dacs, good luck |
14:18.48 | riddlebox | _ShrikE, I have a customer who really wants it and doesnt like idea of not seeing which lines are in use |
14:20.56 | dacs | riddlebox: do you mind if you help me please |
14:21.12 | _ShrikE | riddlebox: If they just want to see what lines are in use, then hints and BLF would work. |
14:21.21 | riddlebox | dacs, I can try |
14:21.41 | riddlebox | _ShrikE, they want to press it and dial and put people on hold |
14:22.28 | dacs | riddlebox: thank you so much, so when i meet with the priest today i will explain to him what i am thinking to do , and how long this project will take! |
14:22.52 | dacs | riddlebox: can you check your pm |
14:23.02 | riddlebox | dacs, it may be two weeks or so for the developer to get it working right |
14:23.44 | riddlebox | _ShrikE, did you see my pastebin? |
14:24.40 | _ShrikE | riddlebox: I missed it, could you paste the link again? |
14:26.18 | riddlebox | sure, let me get it |
14:26.48 | riddlebox | _ShrikE, http://pastebin.ca/924527 |
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14:39.13 | _ShrikE | Looks ok, my experience with SLA is pretty limited though |
14:40.59 | riddlebox | _ShrikE, the SLAStations are your actual extensions right? |
14:45.42 | *** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl) |
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15:15.56 | dacs | trying to config astercrm , the eventsdaemon is asking for my * username and secert...where can i get those , i really forgot them |
15:18.30 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
15:20.35 | kamitodo | anyone familiar with SPA2102? |
15:25.28 | kamitodo | how do I tell if it is locked? |
15:38.46 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
15:45.45 | jameswf-home | pong |
15:46.19 | _ShrikE | kamitodo: have you tried logging in with the info in the manual? |
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15:55.23 | kamitodo | _ShrikE: I have no manual. Bought a used one off amazon. |
15:55.38 | _ShrikE | you can download the manual from linksys |
15:56.50 | kamitodo | looks like both the factory reset and user factory reset command are password protected. does this mean it's locked to a specific ITSP? |
15:57.30 | *** join/#asterisk hi365 (n=hi365@213.151.52.239) |
15:57.36 | jameswf-home | you can usualy crack em open and do a hard reset |
15:58.05 | jameswf-home | if its a vonage box that has connected to the internet it is probably hosed |
16:00.24 | kamitodo | not sure if it's vonage. it makes connections to the following sites: |
16:00.27 | kamitodo | zagbot.com |
16:00.28 | kamitodo | patbox3.patrickdk.com |
16:00.28 | kamitodo | mighty.proclabs.net |
16:00.28 | kamitodo | 64.34.245.224 |
16:00.33 | kamitodo | server.donkeyfly.com |
16:00.36 | kamitodo | www.innomedia.com |
16:00.56 | *** part/#asterisk jivco (n=jivco@85.187.217.6) |
16:02.19 | kamitodo | jameswf-home: what do you mean by crack them open? |
16:03.56 | jameswf-home | like pop out the screws |
16:05.13 | MatBoy | Is there a sipserce out there where I can connect on using my nic cards to test my trunk ? |
16:05.46 | MatBoy | or is this a weird question ? |
16:09.39 | *** join/#asterisk RoyK (n=roy@ti211110a081-7023.bb.online.no) |
16:09.54 | *** join/#asterisk RoyK (n=roy@ti211110a081-7023.bb.online.no) |
16:13.19 | kamitodo | jameswf-home: i see. i'm googling for a how-to on this, but finding nothing... |
16:13.22 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
16:13.28 | *** join/#asterisk lisandropm (n=lisandro@190.1.22.76) |
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16:20.02 | dacs | is there a channel for astercrm |
16:20.12 | dacs | !astercrm |
16:20.25 | MatBoy | dacs, nothing to find on google ? |
16:20.40 | jameswf-home | ~astercrm |
16:20.57 | *** part/#asterisk lisandropm (n=lisandro@190.1.22.76) |
16:20.58 | dacs | MatBoy: i found the webpage but i have a question in the installation part |
16:21.48 | robmac67 | kamitodo: plug in a handset dial **** to access the ivr & then 73738 to do a factory reset |
16:22.00 | MatBoy | dacs, I'm very new her also, I have visited this channel more, but I'm into VoIP now totally... setting up a2billing |
16:22.27 | outtolunc | http://forums.astercrm.org/ |
16:23.34 | dacs | !ivr |
16:23.38 | dacs | ~ivr |
16:23.39 | jbot | [ivr] Interactive Voice Response |
16:23.46 | MatBoy | how do Atcom cards perform, recognized as Digium ? |
16:24.02 | MatBoy | dacs, hehe, I wanted to know that one too :) |
16:24.03 | kamitodo | jameswf-home: i opened the damn thing but there's no obvious reset. |
16:24.22 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
16:24.34 | MatBoy | kaldemar, can' t you look for the chipset how to manually reset it ? |
16:24.40 | MatBoy | sometimes that is possible... |
16:24.45 | MatBoy | hardwired hack |
16:25.16 | dacs | can someone please guide me on where i can install and setup ivr with my * |
16:25.37 | dacs | kaldemar: what are you trying to reset |
16:26.39 | kamitodo | kaldemar? guess you mean kamitodo? I'm trying to reset SPA2102. I don't have the admin pass for it |
16:27.01 | robmac67 | <PROTECTED> |
16:27.20 | kamitodo | did that. there's a password. |
16:27.49 | jameswf-home | jbot: tell dacs about buybook |
16:27.50 | dacs | kamitodo: can you access the web application |
16:28.29 | kamitodo | yes. |
16:29.26 | MatBoy | no-one experience with atcom ? |
16:29.31 | dacs | kamitodo: try user:user |
16:29.43 | dacs | kamitodo: password:8995523 |
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16:31.52 | dacs | kamitodo: is it virgin or SR SPA21202 |
16:32.24 | kamitodo | 8995523 is invalid |
16:32.45 | dacs | ^^ |
16:33.02 | kamitodo | also invalid passwords are the ones posted here: http://www.dslreports.com/forum/remark,14450684~days=9999~start=1900 |
16:33.30 | kamitodo | dacs: how do i tell if its virgin or SR? |
16:34.48 | dacs | kamitodo: is it SPA2102-ACN or SPA2102-R |
16:35.08 | kamitodo | all I know is, upon startup, it's making connections (I'm monitoring via my router) to the following sites: zagbot.com, patbox3.patrickdk.com, mighty.proclabs.net, 64.34.245.224,server.donkeyfly.com, www.innomedia.com |
16:35.59 | kamitodo | the box says spa2102-na, the websites says: SPA-2102 Serial Number:FM500G308331 Software Version:5.1.12 Hardware Version:1.2.5(a) |
16:36.27 | kamitodo | Client Certificate:Installed |
16:36.36 | dacs | kamitodo: get your self a good TFTP software and a proxy server and continue reading that forum, the answer to the quiestion is within |
16:40.34 | drmessano | SPA-2102 with a password? |
16:40.54 | dacs | drmessano: how are you sir |
16:41.06 | drmessano | im ok |
16:41.35 | kamitodo | I'm afraid I'm lost. |
16:42.06 | kamitodo | should I flush the firmware? |
16:42.11 | drmessano | Is it from an ITSP? |
16:42.23 | kamitodo | don't know. bought it used. |
16:42.29 | drmessano | oh |
16:44.19 | kamitodo | if I buy a new one from amazon, are they unlocked? |
16:44.27 | drmessano | Yes |
16:46.10 | [TK]D-Fender | kamitodo, Buy one from a normal reputable dealer and stop asking for trouble |
16:46.39 | drmessano | *NEW* is your friend |
16:47.00 | dacs | [TK]D-Fender: how are you doing sir |
16:47.40 | drmessano | The only NEW Linksys boxes that are questionable are PAP2s, anything else *NEW* from Linksys is going to be (I hate this word) unlocked. |
16:48.09 | dacs | drmessano: unlock, unlock ,unlock |
16:48.29 | dacs | ^^ Don't say UNLOCK ...errrr :) |
16:48.45 | kamitodo | alright. thanks! |
16:49.42 | ManxPower | having a password and being locked are NOT the same |
16:50.19 | kamitodo | btw, since i'm buying new, is the spa2102 the best thing of it's class? |
16:50.43 | kamitodo | ManxPower: ok, what's the diff? |
16:53.19 | drmessano | The difference is whether or not the password or other restrictions are factory defaults |
16:53.53 | drmessano | If some programmed the box via tftp to disallow fact reset and other things, it's not technically "locked" |
16:54.48 | drmessano | I'm not sure it really matters because locked, unlocked, et al are just a bunch of dumbass terms people invented to describe the boxes, and sadly have become common use |
16:55.19 | kamitodo | so if that't the case by flashing again via tftp i can remove the pass |
16:55.55 | drmessano | If the box is set to look at a specific tftp server, then no |
16:56.13 | kamitodo | aha. |
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16:57.09 | drmessano | I suppose you can use a packet sniffer to see where it's looking for it provisioning file |
16:57.17 | drmessano | Then fake a DNS server and a tftp server |
16:57.34 | drmessano | Much like the (oh god) unlocking trick for the PAP2 |
16:58.08 | kamitodo | oh boy. i hoped not to have to do this |
16:58.08 | ManxPower | kamitodo: A locked box can't have it's password reset by a factory default reset |
16:58.26 | *** join/#asterisk RoyK_ (n=roy@ti211110a081-1633.bb.online.no) |
16:59.21 | drmessano | kamitodo: If it you can't fact reset it to even see if it's just provisioned, you're gonna need to do something |
16:59.30 | ManxPower | To me "locked" means, doing a factory reset doesn't clear out all the passwords. Linksys/SIPura don't WANT to lock their boxes, but if they want to sell to the likes of Vonage or other companies that sell the hardware at a loss they must lock them |
17:01.30 | kamitodo | and what does provisioned mean here? bound to a ITSP? |
17:01.35 | drmessano | No |
17:01.43 | ManxPower | I know the universe will explode for saying this but this is one of the best explanations of "locked" I've seen in a long time: "drmessano: The difference is whether or not the password or other restrictions are factory defaults" |
17:01.43 | drmessano | Not per se |
17:02.56 | drmessano | Provisioned is just the act of configuring settings, in this case, via tftp |
17:03.20 | *** part/#asterisk Paladine (n=paladine@ns2.scs-live.com) |
17:03.26 | drmessano | When you start talking about binding, locking, and other bleh terms, those are usually hard coded in |
17:03.59 | ManxPower | There is some special key sequence like ##RESET that clears all setting back to their defaults on an non-locked box. The SIPura docs will have the specific key sequence. |
17:04.31 | drmessano | But if, drmessano, feed a box a file to provision it, and set the field to block fact reset, TO YOU, it's as good as locked until you can reprovision it somehow |
17:05.40 | drmessano | It's called the GPP_K I believe |
17:06.46 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
17:06.47 | kamitodo | ok. how does the packet sniffer help? does the box try to contact their tftp server on startup? |
17:06.57 | drmessano | It could |
17:06.59 | drmessano | Listen.. |
17:07.26 | drmessano | With the PAP2s, they were somewhat easy to (ARRRGH) unlock because there were LOTS of known constants |
17:07.30 | ManxPower | kamitodo: for the amount of time you will spend on trying to hack a locked box you could buy several new ones |
17:07.35 | *** join/#asterisk joelsolanki (i=joelsola@220.224.22.199) |
17:07.47 | drmessano | TFTP addresses, FTP addresses, known GPP_K keys to reset, etc, etc, etc |
17:07.58 | drmessano | You have NO Idea of ANYTHING with this specific box |
17:08.10 | drmessano | Put it back on eBay as "Some used box I had" and move on to a new one |
17:08.40 | drmessano | Even if you DID manage to feed it a xml file.. |
17:08.54 | drmessano | Which the chances of it not needing to be encrypted are slim as hell |
17:09.12 | joelsolanki | Hi All |
17:09.31 | drmessano | You may open it up just enough to release it's got some good fact defaults from an ITSP and was purchased by them in that way |
17:09.37 | drmessano | realize |
17:09.41 | joelsolanki | I am installing zaptel right now. zaptel-1.4.9.2 on centos 5 |
17:09.48 | kamitodo | it won't be ethical to sell it that way. i got it for $20 so no big deal. (no wonder it was cheap!) will just buy a new one. let me know if someone wants the old one, I can just send it to you. |
17:09.53 | joelsolanki | I have installed kernel-devel in centos 5 already |
17:10.11 | *** join/#asterisk angryuser (i=nononon@df01t2-213-44-91-185.d4.club-internet.fr) |
17:10.21 | joelsolanki | i m giving ' make linux26 ' |
17:10.27 | joelsolanki | grep: /lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h: No such file or directory |
17:10.29 | joelsolanki | make: *** No rule to make target `linux26'. Stop. |
17:10.35 | joelsolanki | what could be the problem ? |
17:10.45 | drmessano | Used is used.. Throw it on eBay for $15 and put "ZOMG LQQK THIS IS LOCKED SPA-2102 ** UNLOCKERS SPECIAL **" and let someone else mess with it |
17:10.47 | drmessano | Really |
17:11.32 | joelsolanki | symbolic link is already set |
17:11.33 | joelsolanki | lrwxrwxrwx 1 root root 42 Feb 22 13:17 /lib/modules/2.6.18-8.el5/build -> ../../../usr/src/kernels/2.6.18-8.el5-i686 |
17:11.49 | joelsolanki | still i m not able to install it on centos 5. |
17:11.53 | joelsolanki | any hints pl |
17:11.54 | joelsolanki | pzl |
17:11.56 | ManxPower | joelsolanki: It would not hurt to try installing kernel-source |
17:12.15 | ManxPower | joelsolanki: don't do a make linux26 that is old and not needed |
17:12.36 | ManxPower | unless you are running zaptel 1.0.x or something like that |
17:12.47 | kamitodo | ok, getting a new one. thanks drmessano and everyone else for the advices. |
17:12.50 | drmessano | yum install kernel-devel on CentOS 5 |
17:13.09 | joelsolanki | yes i already install kernel-devel with ym |
17:13.10 | joelsolanki | yum |
17:14.09 | drmessano | Have you updated kernel using something other than yum? |
17:14.40 | ManxPower | joelsolanki: what happens when you just do "make" |
17:14.42 | joelsolanki | just did ' make ' but it is telling that ' you do not appear to have the sources for the 2.6.18-8.el5 kernel installed |
17:15.01 | drmessano | Have you updated your kernel using something other than yum? |
17:15.07 | joelsolanki | let me again try doing yum install kernel-devel |
17:15.15 | *** join/#asterisk tobias (n=tobias@user-0ce2hr3.cable.mindspring.com) |
17:15.21 | drmessano | Have you updated your kernel using something other than yum? ZOMG SHARKS |
17:15.36 | joelsolanki | no i have not. |
17:15.39 | drmessano | Ok |
17:15.40 | joelsolanki | i have used yum only |
17:15.51 | drmessano | Just wanted to make sure you didnt create a version mismatch |
17:16.12 | joelsolanki | hmm |
17:16.22 | joelsolanki | just did yum install kernel-devel |
17:16.24 | joelsolanki | package kernel-devel-2.6.18-53.1.13.el5 (which is newer than kernel-devel-2.6.18-53.1.6.el5) is already installed |
17:16.48 | joelsolanki | is this a problem ? |
17:17.28 | drmessano | Do a yum list kernel* |
17:17.33 | joelsolanki | ok |
17:18.01 | drmessano | Make sure the 2.6.XX-XX are the same all the way down |
17:18.07 | joelsolanki | kernel.i686 2.6.18-8.el5 installed |
17:18.11 | joelsolanki | kernel.i686 2.6.18-53.1.13.el5 updates |
17:18.53 | drmessano | sounds like you need to update the kernel to match the sources |
17:19.03 | joelsolanki | oh. |
17:19.17 | joelsolanki | yum update kernel* ?? |
17:19.32 | drmessano | yeah |
17:19.38 | joelsolanki | let me do that |
17:19.49 | drmessano | I'm not 100% sure about that.. but I have had this problem with that solution |
17:20.03 | joelsolanki | got it |
17:21.41 | joelsolanki | i think i have to reboot the server one time ? |
17:21.49 | joelsolanki | kernel is installed therefore |
17:21.50 | drmessano | I'm headed out for a bit.. I wanted to put my $.02 in about the kernel versions because i've gotten them out of sync before and had issues like yours.. i'm sure if that's not the ONLY problem, someone else a heck of lot better with Zaptel can help ya |
17:21.56 | drmessano | Yeah you do |
17:22.00 | joelsolanki | ok |
17:22.15 | drmessano | ttfn |
17:22.23 | joelsolanki | ok |
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17:25.33 | *** join/#asterisk puzzled (n=patrick@53533DDB.cable.casema.nl) |
17:28.06 | joelsolanki | ok updating kernel worked !! |
17:28.08 | joelsolanki | :) |
17:29.48 | joelsolanki | googling on ss7 found that chan_ss7 is the best to use open source though they have stopped the development. |
17:30.23 | joelsolanki | I have sangoma A104D. working on integrating sangoma A104D + ss7 + asterisk |
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17:56.22 | TJNII | What do the sip Parse_srv messages mean? I'm not getting very far with google. |
17:57.09 | TJNII | DNS? |
17:57.54 | *** join/#asterisk tobias (n=tobias@cpe-066-026-085-055.nc.res.rr.com) |
17:59.06 | ManxPower | SRV is a DNS method to automagically find the correct IP address for services (like SIP) for a domain |
17:59.19 | ManxPower | srvlookup=no in sip.conf |
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18:00.25 | TJNII | Aah, ok. That's what I gleaned from the source. |
18:00.40 | TJNII | It's not causing problems, I just didn't know what the messages meant. |
18:02.36 | ManxPower | it might show down connections if you don't need it. |
18:13.01 | lnx | ho there, do you know solotion for ringing detection in AGI with perl? I want to create ,if ringing hangup, like script |
18:14.09 | ManxPower | lnx: I don't believe there is a way. |
18:14.47 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
18:15.10 | JunK-Y | lnx: ur AGI must listen on the AMI port and listen to events. |
18:15.27 | jmacz | Hi everyone, I'm having dropped incoming calls (27 hangup_cause on PRI debug). We've checked the line twice with the Telco guys and it's physically and data-link OK. If we connect the old PBX, none of the incoming calls are dropped. Any ideas what may be causing this behavior (we are using a te210P)??? |
18:15.32 | ManxPower | of course there are many kinds of "ringing" |
18:16.26 | ManxPower | Cause 27 is Destination Out of Order |
18:17.00 | lnx | ManxPower JunK-Y okay ty. Where can i fond more info about it? the-asterisk-book.com? |
18:17.05 | ManxPower | Maybe your have the wrong switchtype= You could expect national, 5ess, and DMS to mostly work for any swotch |
18:17.07 | jmacz | ManxPower, that's it, and it may be caused by physical data-link problems, right? |
18:17.15 | ManxPower | lnx: We don't even know what type of ringing you are talking about |
18:17.24 | ManxPower | jmacz: I've never seen that on INCOMING calls. |
18:17.36 | jmacz | ManxPower, it's EuroISDN, and it's the only one I guess that fits our Telco |
18:17.58 | lnx | ManxPower: well i have defined a callfile with channel asterisk makes a simple call ... |
18:18.18 | *** join/#asterisk steliosk (n=Stelios@79.131.73.109) |
18:18.57 | jmacz | ManxPower, we've seen with the Telco guys that we send a PROGRESS and the receive an STATUS before the Telco sends us the DISCONNECT with the 27 hangup cause |
18:18.59 | lnx | ManxPower: if the called endpoint is ringing ... |
18:19.17 | ManxPower | lnx: Have the .call file send the call to Local/ then in the dialplan use a Dial with a very short timeout, then check the value of HANGUPCAUSE or DIALSTATUS |
18:20.06 | lnx | ManxPower: thank you |
18:20.10 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
18:20.29 | ManxPower | lnx: none of this will work on FXO (analog or T-1) signaled port. |
18:20.44 | lnx | ManxPower: how can i get DIALSTATUS variable? |
18:20.59 | ManxPower | ${DIALSTATUS} |
18:21.10 | ManxPower | NONE of this requires AGI |
18:21.27 | ManxPower | except for maybe the creation of the .call file |
18:21.27 | lnx | i see , thx |
18:22.31 | *** join/#asterisk tris (i=tristan@camel.ethereal.net) |
18:22.42 | jmacz | ManxPower, may it be a chan_zap issue? |
18:23.44 | ManxPower | jmacz: I have no idea. |
18:24.17 | ManxPower | I would do a pri debug or pri intense debug, report it to bugs.digium.com. Make sure there is information from ONLY one failed call. |
18:24.46 | jmacz | ManxPower, ok, I'll try that |
18:24.54 | jmacz | thank you very much |
18:34.29 | *** part/#asterisk kamitodo (n=user@70.17.85.123) |
18:35.16 | *** join/#asterisk nvrpunk (n=root@81.90.21.227) |
18:35.54 | nvrpunk | I have a portech gsm 372, its working fine outbound now I am trying to get inbound working |
18:35.58 | nvrpunk | I get -- SIP/1011-081b6030 is making progress passing it to SIP/sip10-0826c010 |
18:36.08 | nvrpunk | when trying to call to it |
18:36.30 | nvrpunk | i want it to route to 1012 which it is configured to do |
18:36.34 | nvrpunk | but all I get is ringing |
18:36.44 | nvrpunk | and the device doesnt actually monitor its call progress |
18:37.01 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
18:38.34 | MatBoy | is it possible to call internal without a trunk ? |
18:39.05 | MatBoy | so only on the LAN withotu having a trunk for this to the outside world ? |
18:40.07 | ManxPower | ~trunk |
18:40.08 | jbot | somebody said trunk was is a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
18:40.18 | ManxPower | We don't use the word trunk here. Use the right term? |
18:40.51 | *** part/#asterisk bkw_ (n=brian@adsl-70-234-169-126.dsl.tul2ok.sbcglobal.net) |
18:41.01 | MatBoy | ManxPower, I used that term because I'm using a2billing and I have the idea that I need a " trunk" to be able to make a call anyhow |
18:41.54 | TJNII | You'd probably get a better answer if you describe in detail what you are trying to do. |
18:42.09 | MatBoy | ok, if that's allright I will do that |
18:42.12 | ManxPower | we don't really support billing software here. |
18:42.15 | MatBoy | I don |
18:42.25 | MatBoy | I don't want to be annoying :) |
18:43.45 | MatBoy | I have ordered a Digium E1 card that is not here yet, so what I try to do is calling on LAN to test it already when I setup 2 customers and call as " friend" in the system using a prefix for internal calls |
18:44.10 | MatBoy | my question is if this will be possible so I'm not looking or trying to do something that can;t be done |
18:44.52 | ManxPower | MatBoy: SIP? |
18:44.54 | drmessano | ~book |
18:44.54 | 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 |
18:45.07 | MatBoy | ManxPower, yes |
18:45.43 | ManxPower | MatBoy: you would edit sip.conf to set up the SIP peers and devices, then edit extensions.conf to set up the call routing. What SIP phone are you using? |
18:45.46 | MatBoy | yep, I'm looking at every known document of course, I'm not a "ask for a solution" person :) |
18:46.14 | MatBoy | ManxPower, x-lite to test at the moment on 2 seperate computers |
18:46.48 | *** join/#asterisk ZPertee (n=ZPertee@cpe-98-27-248-172.neo.res.rr.com) |
18:47.43 | MatBoy | ManxPower, but the actuall question is maybe, will a2billing not manage this by default or do I always have to configure asterisk also for the a2billing part ? I mean, fir a2billing you need to include config files into the asterisk config files |
18:48.11 | ManxPower | MatBoy: we don't know ANYTHING about a2billing here. We deal with Asterisk, not 3rd party billing systems |
18:48.22 | ManxPower | I suggest you use the correct a2billing support options |
18:49.02 | MatBoy | ManxPower, I understand, but I was pointed to a2billing here some time ago, so I assumed people were using it also ? Just some global knowledge sharing ;) |
18:49.24 | MatBoy | ManxPower, I'm already crawling their forum and docs a lot |
18:50.02 | ManxPower | MatBoy: Best of luck |
18:50.57 | MatBoy | ManxPower, yep, thanks ! but I don't want to act like I'm looking for a solution, only the right way where I might be question myself what direction I should take.. need some confirment in the beginning sometimes :) |
18:51.04 | MatBoy | I hope you know what I mean |
18:51.44 | ManxPower | I think what you mean is that you are finding the a2billing support to be lacking so you are here in a desperate attempt to get help. |
18:52.06 | MatBoy | ManxPower, no, not exactly |
18:52.14 | drmessano | MatBoy: It sounds to me like you seriously need to work on your understanding of Asterisk before even attempting something like A2Billing which is going to require a LOT of assumptions due to its lack of documentation |
18:53.34 | MatBoy | ManxPower, I'm trying to tell that if it's not allowed here to talk about 3rd party, I don't have a problem with it but just a question. Why not share info that is not 100% related when someone has it.. I almost do the same on other channels like vmware.. if someone has a HW question... why not ? |
18:54.06 | MatBoy | drmessano, yes there is missing some stuff indeed, but still, you need to start somewhere... |
18:54.12 | drmessano | Yes you do |
18:54.13 | drmessano | The book |
18:54.15 | drmessano | ~book |
18:54.15 | 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 |
18:54.34 | MatBoy | drmessano, yep, found that one yesterday, very nice 600 pages :) |
18:54.37 | MatBoy | good resource |
18:54.40 | drmessano | This channel isn't for "I just installed Asterisk, help me please callcenter".. This is more for intermediate issues |
18:54.44 | drmessano | Ok good, go read |
18:55.15 | MatBoy | drmessano, that is what I wanted to say... I'm not that kind of guy... I will be here all the time when I'm online... I can share my knowledge later on also... |
18:55.19 | drmessano | After you have read, maybe some of the poorly documented pieces of A2billing will make sense |
18:55.37 | drmessano | MatBoy: You are sounding exactly the opposite of what you are saying |
18:55.46 | MatBoy | drmessano, why would I be here all day already than ;) I should have quite already when my question was not answered :) |
18:55.50 | drmessano | "I am not the kind to not read and beg for help, but can you help me please" |
18:56.02 | MatBoy | drmessano, that's because I don't want to be rude, I hate those people also |
18:56.14 | drmessano | Great.. So don't be one.. |
18:56.17 | MatBoy | drmessano, did I beg somewhere ? |
18:56.20 | ManxPower | MatBoy: you are trying to amputate a diseased limb without knowing basic anatomy |
18:56.26 | jameswf-home | ~rude |
18:56.26 | jbot | rude is, like, making me tell people things that they dont want to be told |
18:56.26 | TJNII | MatBoy: Have you played with just a basic * server yet? No billing stuff, just a virgin install? |
18:56.57 | drmessano | You were arguing with ManxPower over asking a question about an unsupported app and you clearly haven't got past Asterisk 101 yet |
18:57.20 | jameswf-home | heh I can amputate a limb without basic anatomy as long as there are no expectations after the deed is done |
18:57.20 | drmessano | You're only making this harder on yourself |
18:57.24 | MatBoy | TJNII, before a little bit, but I'm actually discovering how such software would be a good additional thing to *, because when it isn't... you will think you can do stuff with it what never could be done. |
18:57.25 | drmessano | LOL |
18:57.45 | drmessano | ~nowwhat |
18:57.46 | jbot | So you just installed asterisk and arent sure what to do now? visit http://www.a1b2c3.com/suilodge/metfun1.htm |
18:57.56 | drmessano | ~now what |
18:57.57 | jbot | i guess now what is 2*4? |
18:57.57 | drmessano | crap.. where is it |
18:58.09 | jameswf-home | jbot spank MatBoy |
18:58.09 | jbot | ACTION bends MatBoy over his knee and tatoos 'ibot' on MatBoy's pasty white buttocks. |
18:58.14 | MatBoy | drmessano, I think you like the bot ? |
18:58.33 | *** part/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
18:58.40 | MatBoy | drmessano, the simple question is... did you already know everything ? I guess not and I'm not saying I know anything at all |
18:58.59 | drmessano | You're asking about an app and you haven't gotten past Asterisk 101 yet |
18:59.05 | drmessano | GO READ and stop ARGUING YOUR POINT |
18:59.05 | jameswf-home | Between me, jbot, and google I know everything |
18:59.30 | drmessano | You *ARE* being one of those annoying guys by continuing this line of insane arguing |
18:59.37 | jameswf-home | ~troll |
18:59.37 | jbot | hmm... troll is a race on some muds; a guy under a bridge: an annoying robot; a port scanner, or somebody faking being clueless to be shown the One Linux Way so he can argue against it for his psychology thesis on linux advocates, or an employee of trolltech, or at http://www.kuro5hin.org/story/2001/7/27/51233/2979 or ... |
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18:59.43 | drmessano | Have you even gotten two extensions calling each other yet? |
18:59.49 | MatBoy | drmessano, yes I will, but as you also have seen, a lot of people think it's simple to setup, and I have seen it's not... so I'm discovering what will be best |
18:59.59 | nvrpunk | what is the best way to encode wave files to g729? |
19:00.08 | jameswf-home | simple is a relitive term |
19:00.19 | drmessano | MatBoy: It will be simple after you.. ummm |
19:00.28 | drmessano | Read? |
19:00.29 | jameswf-home | rtfm |
19:00.39 | MatBoy | drmessano, ofcoruse, did I say it wasn't ? |
19:00.46 | MatBoy | no ;) |
19:00.52 | drmessano | Dude, stop trolling.. |
19:01.17 | MatBoy | I'm not your dude, please stop trolling also... I think we have solved this than |
19:01.26 | drmessano | Go back to your bridge |
19:01.40 | drmessano | With a copy of the book |
19:01.55 | jameswf-home | ~dude |
19:01.56 | jbot | Be most excellent to each other! |
19:02.06 | drmessano | Tried to help and now, alas, I must terminate this communication |
19:02.29 | MatBoy | drmessano, don't tell me what to do, just give advice or don't say anything... that's looking like a wiseguy actually... sorry, but you are right, but the way you say is very.... uhm ;) |
19:02.35 | TJNII | MatBoy: You don't actually have a problem. You just don't understand how asterisk works. Read the book and play with a clean * install. You're not ready for a2billing. |
19:02.40 | jameswf-home | MatBoy: did you try rebooting |
19:02.59 | MatBoy | TJNII, yes I will continue :) |
19:04.43 | MatBoy | TJNII, the book is very good... but the a2bliing part let it look likes it's automated.. like hosters that think they can admin servers when they can install PLesk.... but I wanted to know for sure if that's not it, because I know for sure it isn't :) |
19:04.50 | drmessano | :)))))) (((((((((: ;) |
19:06.30 | jameswf-home | simheh |
19:06.37 | jameswf-home | ~fixit |
19:06.37 | jbot | to fix your issue follow these five steps... 1. Find a radio with a long cord, use an extension cord if needed to get a plug without a GFI. 2. Plug in the radio to the non GFI outlet. 3. Fill your bathtub with water. 4 bring radio with you and step into the tub 5. drop said radio. PROBLEMS SOLVED |
19:07.40 | MatBoy | jameswf-home, I think you never looked at discovery channel... that one was busted ;) |
19:07.46 | drmessano | Ok, back out to finish off my Magic Mystery Tour saturday.. I think the wife and I will stop at the A2billing store at the mall and get a book on pretzels |
19:08.14 | drmessano | Chowder |
19:09.01 | jameswf-home | Lies all Lies get a video camera and show me |
19:09.45 | MatBoy | jameswf-home, I think it;s somewhere on youtube if you want... water does not do such things very easy with power, it's much more a resistor |
19:10.13 | jameswf-home | I saw the episode I believe it was a pool |
19:10.51 | MatBoy | jameswf-home, what I remember it was a bath with a multimeter and some stuff where they let a radio fall into the water from a shelf |
19:11.15 | MatBoy | very simple explained |
19:11.27 | MatBoy | I never saw one with a pool, I don't stay home for it |
19:11.42 | scooby2 | fsck a duck. Well its not my digium te212p as this sangoma a102 errors the same way. All zap calls stop. Errors like this: chan_zap.c: Ring requested on channel 0/2 already in use or previously requested on span 2. Attempting to renegotiating channel. |
19:12.07 | *** join/#asterisk Vorbote (n=vorbote@unaffiliated/vorbote) |
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19:13.19 | ZPertee | I have an avaya pbx that I want to use with *. What I want to do is connect an extension port on the avaya to my digium fxo card. however that isn't working. what signalling should I use? |
19:14.43 | jameswf-home | ok Episode 22 Bathtub Electrocution: CONFIRMED - virtually everything they dropped in the tub registered as a fatal shock |
19:15.50 | jameswf-home | who stays up to watch tv I tivo and catch up saturday |
19:18.08 | *** join/#asterisk Greek-Boy (n=email@41.221.58.4) |
19:18.40 | MatBoy | jameswf-home, ah ok, I thought the shock was to low, there was something they discovered that they didn't expect, that's what I remember |
19:19.11 | MatBoy | jameswf-home, what I know is that with a very simple and low shock you can mess up the rythm of you hard indeed |
19:19.12 | jameswf-home | I think you can drop a radio into distilled water and live... |
19:19.31 | MatBoy | jameswf-home, let's ask them to test that comparing to this test |
19:19.38 | MatBoy | would be nice to see |
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19:22.49 | *** join/#asterisk pkwong (n=chatzill@server.virtutel.net) |
19:23.03 | pkwong | hi all. |
19:23.22 | jameswf-home | holy speak of the devil batman |
19:23.29 | pkwong | hehe. |
19:23.33 | pkwong | hey james |
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19:39.01 | outtolunc | .. |
19:41.02 | nvrpunk | for an incoming number of say 07906824351@1012 |
19:41.15 | nvrpunk | _1012 should catch it correct? |
19:41.53 | outtolunc | 1012 is a [1012] not an exten |
19:42.26 | outtolunc | @ domain/context |
19:42.34 | nvrpunk | ok |
19:45.22 | *** join/#asterisk m4sk4r4 (n=m4sk4r4@189.13.188.14) |
19:46.56 | tzafrir_home | jameswf-home, ping |
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20:00.13 | nvrpunk | i cant get this portech gdm372 to do incoming calls -.- |
20:00.17 | nvrpunk | no matter how hard i try |
20:00.20 | dssman | <PROTECTED> |
20:00.39 | nvrpunk | means its not finding the asterisk box |
20:00.43 | nvrpunk | is what it sounds like |
20:00.51 | outtolunc | look on the floor, is the cable laying there <G> |
20:02.32 | dssman | lol |
20:02.54 | nvrpunk | so I add exten => _1012,1, Dial(SIP/sip1) for the portech gsm372 and it just rings and rings |
20:02.58 | dssman | its server side I see the error... I can dial extensions, just the phone isnt registering one itself |
20:03.14 | nvrpunk | even though the portech is showing 0790blah@1012 |
20:03.22 | nvrpunk | as incoming to it |
20:06.28 | nvrpunk | check your configs |
20:07.15 | dssman | if Im using asteriskNow, and I add a user, will that add extensions for me too? |
20:07.31 | *** join/#asterisk rbd (n=rbd@216.148.216.55) |
20:07.38 | linuxstb | Hi all. I've set up a basic asterisk 1.4.18 system (hosted on a server located in a datacentre), and have various different kinds of SIP phones (Grandstream GXP-2000, a SPA3102, Ekiga softphone and Nokia E60) attached to my home network (behind a NAT firewall) for testing purposes. |
20:07.41 | linuxstb | They can all dial each other, apart from the Nokia, which can make outgoing calls to the other phones, but refuses to accept any incoming calls - it never rings and seems to crash the Nokia SIP stack (I need to reboot the phone afterwards). |
20:08.53 | rbd | hi guys, I had a question about the stat function. the wiki docs on it are unclear. if I call stat(e, ...) for instance, what is returned if the file exists, and if it doesnt? |
20:11.07 | jmacz | dssman, check with *CLI> sip show peer <your_phone> if it's defined somewhere (like sip.conf) |
20:13.53 | dssman | yea appears to b |
20:14.46 | dssman | I hate waiting for the phone to reboot :P |
20:15.38 | dssman | okay, gettin closer... now I have username / auth mismatch |
20:15.41 | *** join/#asterisk SteveTotaro (n=root@pool-71-179-207-15.bltmmd.east.verizon.net) |
20:15.56 | dssman | what is digest? |
20:16.43 | dssman | and for every phone that is going to be connected do I need ot make an entry in the sip and exten .conf files? |
20:17.11 | dssman | OMG I have a registered phone |
20:18.25 | dssman | uch, now none of the demo extensions are working |
20:18.27 | dssman | grr |
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20:23.11 | *** join/#asterisk [Gandhi] (n=[Gandhi]@201.211.12.64) |
20:23.35 | dssman | ohh, lol nothin works :D |
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20:39.16 | TJNII | dssman: You have a registered phone? |
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20:41.18 | teknoprep | i love polycom phones |
20:42.10 | drmessano | I love little chickens and shiny things |
20:43.17 | teknoprep | went on ebay and purchased a bunch of cheap ip 501's |
20:44.22 | drmessano | The famous shoretel phones? |
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20:49.04 | *** join/#asterisk SparFux (n=raoul@e182029172.adsl.alicedsl.de) |
20:49.18 | TJNII | I can never find polycoms cheap. |
20:49.53 | SparFux | I have problems compiling mISDN for AMD64 on debian. |
20:50.55 | SparFux | mISDN-1_1_7_2 doesn't compile. It says something needs to get fixed from CFLAGS to EXTRA_CFLAGS. And the git version compiles, but when activating capi it crashes. The whole system freezes. |
20:51.37 | TJNII | That doesn't sound like a compile problem. |
20:53.34 | SparFux | With the git version, it is not. |
20:53.46 | SparFux | I don't know, wether it has something to do with AMD64 arch. |
20:54.04 | drmessano | Always safer to go with 32-bit |
20:55.41 | SparFux | depends. The NXbit is a nice feature busting almost all buffer overflow issues. |
20:56.13 | drmessano | Features are great when some parts don't work well under 64-bit |
20:57.17 | SparFux | yet |
20:57.33 | drmessano | I hate that word |
20:57.44 | SparFux | neway, I prefer the safe thing with some thing not working over the unsafe thing. |
20:58.18 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
20:59.08 | drmessano | I think it's dumb buying 64-bit hardware for Asterisk when the same money can get you a decent multicore 32-bit and some expectation that various drivers for hardware will all work |
20:59.18 | drmessano | But thats my personal opinion.. |
21:02.01 | SparFux | Well, I just switched from my 32 bit system over to this 64 bit system. |
21:02.06 | SparFux | right today :-) |
21:02.41 | SparFux | As long as SMP is in place, I will refuse to buy multicore. SMP is a bad idea. |
21:03.00 | SparFux | And even worse is 8-core cpus. |
21:03.41 | drmessano | Heh.. and buying a 64-bit cause "One day it will all work" is better? :) |
21:03.54 | SparFux | I think it will work in one day. |
21:03.59 | SparFux | exatly in one day :-) |
21:04.05 | lnx | i have no idea how can i make a procedure does: call a number via callfile; if the endpoint is ringing hang up; else do blahh. |
21:04.07 | SparFux | I am not even sure it is a 64 bit issue. |
21:04.22 | riddlebox | drmessano, have you ever dealt with sla? |
21:04.29 | *** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6) |
21:04.35 | SparFux | plus, it is the git version, misdn 1.2. It is not supposed to be stable. |
21:04.37 | drmessano | SLA is evil, no |
21:04.52 | riddlebox | why is it evil? |
21:05.26 | drmessano | I just don't believe in SLA or the Easter Bunny |
21:05.51 | *** join/#asterisk seanbright (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
21:06.15 | lnx | ring detection is impossibble :/ |
21:09.05 | jameswf-home | impossible for who |
21:11.25 | SparFux | Get this: http://linuxgazette.net/107/pramode.html |
21:12.16 | *** join/#asterisk puzzled (n=patrick@53533DDB.cable.casema.nl) |
21:12.51 | lnx | jameswf-home: which variable contains this : logger.c: -- SIP/0009*001-0959fb30 is ringing ? |
21:14.13 | jameswf-home | likely verbose.... |
21:14.23 | jameswf-home | couldnt say without looking |
21:14.30 | SparFux | I get this strange error when compiling: scripts/Makefile.build:46: *** CFLAGS was changed in "/usr/src/mISDN/mISDN-1_1_7_2/drivers/isdn/hardware/mISDN/Makefile". Fix it to use EXTRA_CFLAGS. Stop. |
21:14.53 | [TK]D-Fender | lnx, youare going to have to complete recode your own dial process..... goo luck with that... |
21:15.52 | seanbright-home | SparFux: what do the mISDN maintainers say? |
21:18.43 | lnx | [TK]D-Fender: what do you think, why asterisk hasn't a ringing status variable? BTW i can't recode. |
21:19.12 | seanbright-home | SparFux: and didn't you already say that the git version works? why are you using the release version? |
21:19.35 | SparFux | git version crashes when activating capi. |
21:19.46 | seanbright-home | SparFux: ahh |
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21:20.39 | tzafrir_home | SparFux, is CLAGS used explicitly there? |
21:20.53 | SparFux | Yes. It is. |
21:21.09 | lnx | it is weird there is no solution for call testing what analyse ringing only :> |
21:21.15 | tzafrir_home | Adn you fixed it to use CFLAGS? |
21:21.59 | *** part/#asterisk Vorbote (n=vorbote@unaffiliated/vorbote) |
21:22.54 | lnx | jameswf-home: i mean variable like DIALSTAUTS. |
21:23.01 | lnx | *STATUS |
21:23.33 | [TK]D-Fender | lnx, you can't do anything in the middle of a DIAL. |
21:24.02 | lnx | saaad :( |
21:24.28 | seanbright-home | lnx: i missed the beginning of the question, what are you trying to accomplish exactly? |
21:24.29 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
21:24.49 | adeel | is there a way to disable showing ALL manager connects? |
21:24.54 | adeel | outside of hacking the source? |
21:25.41 | seanbright-home | i doubt it |
21:25.49 | lnx | seanbright-home: main :) i have no idea how can i make a procedure does: call a number via callfile; if the endpoint is ringing hang up; else do blahh. |
21:26.24 | lnx | there is an entry in full.log logger.c: -- SIP/0009*001-0959fb30 is ringing |
21:26.40 | *** join/#asterisk Quaver (n=Onyxyte@r74-192-234-77.lfkncmta01.lfkntx.tl.dh.suddenlink.net) |
21:26.50 | seanbright-home | lnx: you would have to modify the Dial application |
21:27.14 | seanbright-home | lnx: once you are in Dial, you can't do anything until the call leaves the application |
21:27.18 | lnx | may a variable like DIALSTATUS contains * is ringing |
21:27.41 | seanbright-home | lnx: no, unfortunately there is no "out of the box" way of doing that |
21:27.50 | seanbright-home | lnx: you'll have to modify the source or have someone do it for you |
21:27.50 | lnx | damn |
21:28.09 | seanbright-home | adeel: what version are you running? |
21:28.11 | *** join/#asterisk cesar_CR (n=cesar@celord.ice.co.cr) |
21:28.26 | [TK]D-Fender | lnx, Why do you want to hang up if its rining? |
21:28.46 | adeel | 1.4.8 |
21:29.10 | anonymouz666 | [TK]D-Fender, just curious, do you know if it's possible to 'reinject' a PRIO do caller into queue, once the caller is already waiting? |
21:29.24 | *** join/#asterisk teknoprep (n=teknopre@unaffiliated/teknoprep) |
21:29.38 | [TK]D-Fender | anonymouz666, nothing I'm aware of |
21:29.40 | lnx | [TK]D-Fender: because i have to run test calls ... |
21:29.41 | adeel | seanbright-home, i set displaymanagerconnects = no in manager.conf but that doesn't seem to be doing anything |
21:30.26 | [TK]D-Fender | lnx, Sorry, unless you're prepared to do some serious recoding your idea jsut isn't workable |
21:30.29 | lnx | if u have an idea how to test automatically thet calls are works fine |
21:30.38 | lnx | please share with me :) |
21:30.50 | seanbright-home | adeel: try 'displayconnects = no' |
21:31.11 | adeel | seanbright-home, i'll try |
21:31.18 | lnx | *that |
21:31.23 | seanbright-home | adeel: make sure to come back and thank me |
21:31.27 | seanbright-home | ;-) |
21:31.52 | drmessano | If it doesn't work, will you give him a Twinkie? |
21:32.08 | seanbright-home | and it will work |
21:32.15 | seanbright-home | i'll bet my last twinkie on it |
21:32.17 | seanbright-home | ... |
21:32.21 | drmessano | Says he who has no Twinkies |
21:32.38 | lnx | i'm using Twinkle :) |
21:33.32 | adeel | seanbright-home, well now it just keeps saying == Parsing '/etc/asterisk/manager.conf'; Found |
21:33.38 | *** join/#asterisk _matt (i=matt@2001:770:168:1:20b:cdff:fe04:843a) |
21:33.45 | seanbright-home | adeel: but the connects are gone, yeah? |
21:33.50 | adeel | seanbright-home, yes |
21:33.53 | adeel | seanbright-home, so i guess thanks |
21:33.57 | seanbright-home | heh |
21:34.01 | drmessano | Ohhhh |
21:34.11 | drmessano | That's a HALF-TWINKIE solution |
21:34.14 | adeel | haha |
21:34.22 | seanbright-home | the "Parsing" foo isn't coming from the manager, its coming from the configuration system |
21:34.39 | adeel | anyway to disable that? |
21:34.51 | seanbright-home | set verbose to 1 |
21:34.58 | seanbright-home | err |
21:35.02 | seanbright-home | turn off verbose |
21:35.04 | seanbright-home | :) |
21:37.08 | seanbright-home | lnx: with the test you are building, what is the alternative to ringing? |
21:37.15 | adeel | seanbright-home, ah, well i need the verbose level when i'm debugging |
21:37.22 | seanbright-home | adeel: then you're screwed. |
21:37.49 | adeel | seanbright-home, yeah i know...i wonder if there's a way to move the == Parsing statements to a different logging level/context ....e.g. debug or something |
21:37.51 | seanbright-home | adeel: or delete line 826 of main/config.c |
21:38.18 | adeel | line 826 huh? i'll take a look at it |
21:38.18 | adeel | thanks |
21:38.20 | seanbright-home | adeel: change 826 from: |
21:38.21 | seanbright-home | ast_verbose(VERBOSE_PREFIX_2 "Parsing '%s': ", fn); |
21:38.23 | seanbright-home | to: |
21:38.38 | adeel | now if only i could get call pickup working right.... |
21:38.43 | seanbright-home | ast_log(LOG_DEBUG, "Parsing '%s': ", fn); |
21:38.51 | seanbright-home | and you're golden. |
21:39.09 | adeel | seanbright-home, oo, very interesting |
21:39.29 | seanbright-home | then create a patch file so you can do it again when you upgrade |
21:39.58 | adeel | yeah, i have to create a patch file anyways...it's easier using my package manager (i'm on gentoo) |
21:40.09 | seanbright-home | gotcha |
21:40.20 | seanbright-home | well good luck and god speed. |
21:40.36 | *** join/#asterisk ManxPower (n=manxpowe@ip72-204-168-61.no.no.cox.net) |
21:40.53 | drmessano | ROFL... |
21:40.54 | adeel | seanbright-home, any experience with call pickup? |
21:41.04 | seanbright-home | adeel: no sir |
21:41.12 | drmessano | Im watching the CLI as a buddy of mine is getting his wakeup call for work |
21:41.22 | drmessano | He keeps failing the math question from Allison |
21:41.31 | seanbright-home | there's a math question? |
21:41.33 | adeel | what's the question? |
21:41.36 | seanbright-home | is that to ensure you're actually awake? |
21:41.50 | drmessano | Yes |
21:41.58 | drmessano | I have a somewhat modified version |
21:42.13 | drmessano | -- Playing 'wrong-try-again-smarty' (escape_digits=) (sample_offset 0) |
21:42.16 | lnx | seanbright-home: i'm out sorry |
21:42.23 | lnx | go to sleep :P |
21:42.37 | lnx | thanks the help |
21:43.13 | *** join/#asterisk ectospasm (n=ectospas@c-71-207-229-248.hsd1.al.comcast.net) |
21:43.46 | seanbright-home | lnx: if all you care about is "failure" |
21:43.54 | seanbright-home | you could just set a short timeout on the dial |
21:43.59 | seanbright-home | like, say, 5 seconds |
21:44.19 | drmessano | Hmm |
21:44.31 | seanbright-home | then check DIALSTATUS when you get back from Dial(), and if its CANCEL then it timed out |
21:44.43 | seanbright-home | otherwise you could assume it rang |
21:45.18 | seanbright-home | its also possible that wouldn't even come close to being correct |
21:45.20 | seanbright-home | but its a thought. |
21:46.14 | drmessano | ROFL |
21:46.18 | drmessano | wow |
21:46.32 | drmessano | I think he left his phone on Auto-Answer.... |
21:47.07 | drmessano | He's been on the phone with Allison for 10hours and 20minutes |
21:47.10 | nvrpunk | can rasterisk convert all files in a directory? |
21:47.27 | nvrpunk | with one command |
21:47.34 | hi365_m | any idea what this error means? |
21:47.34 | hi365_m | [Mar 1 23:46:16] NOTICE[3791] chan_sip.c: Unable to create/find SIP channel for this INVITE |
21:47.34 | hi365_m | [Mar 1 23:46:16] WARNING[3791] chan_sip.c: sip_xmit of 0xeab194 (len 486) to 212.150.88.20:5060 returned -2: Network is unreachable |
21:47.55 | nvrpunk | reinvite=no |
21:47.56 | seanbright-home | nvrpunk: no sir |
21:48.16 | nvrpunk | seanbright-home, anyone mind helping me with this script then? for I in *.wav; do convert -OPTIONS "$I" "$I".g729; done |
21:48.17 | seanbright-home | nvrpunk: can't use sox? |
21:48.21 | jql | for i in dir/*; do rasterisk ...; done |
21:48.24 | jql | voila |
21:48.25 | nvrpunk | does sox do g729? |
21:48.28 | jameswf-home | my sox are dirty |
21:48.35 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
21:49.03 | jql | well; do rasterisk ... $i; done |
21:49.40 | drmessano | I scrwed up |
21:49.46 | *** join/#asterisk amir_ (n=amir@unaffiliated/amir) |
21:49.51 | drmessano | I did chan_spy, but I should have transferred the call |
21:49.52 | seanbright-home | nvrpunk: what is 'convert'? never used it |
21:50.05 | nvrpunk | seanbright-home, convert = rasterisk it was like foo |
21:50.11 | *** part/#asterisk amir_ (n=amir@unaffiliated/amir) |
21:50.13 | seanbright-home | ohhh |
21:50.20 | nvrpunk | just mean convert using whatever tool |
21:50.35 | *** join/#asterisk vrtk (n=bruno@201009059057.user.veloxzone.com.br) |
21:51.41 | seanbright-home | for a in /path/to/files/*.wav; do asterisk -rx "file convert $a `echo $a | sed -e s/\.wav$/.g729/g`"; done |
21:51.45 | seanbright-home | that might work |
21:51.57 | seanbright-home | and i emphasize the word _might_ |
21:52.13 | hi365_m | nvrpunk:was that for me (reinvite-no)? |
21:52.31 | nvrpunk | hi365, yes |
21:52.57 | riddlebox | weird, I upgraded to 1.4.18, and now it seems that the system is seeing the sla stuff |
21:53.33 | nvrpunk | for i in *; do rasterisk -x "$i" "$i".g729"; done |
21:53.33 | nvrpunk | <PROTECTED> |
21:53.42 | seanbright-home | ? |
21:53.48 | seanbright-home | what was unclear about: |
21:53.49 | seanbright-home | for a in /path/to/files/*.wav; do asterisk -rx "file convert $a `echo $a | sed -e s/\.wav$/.g729/g`"; done |
21:54.26 | jameswf-home | escape the spaces |
21:54.27 | hi365_m | nvrpunk: didnt work |
21:54.41 | nvrpunk | hi365, no clue then |
21:54.43 | hi365_m | can having the hosname settings wrong have anything to do with that error? |
21:56.41 | nvrpunk | seanbright-home, that gives me unable to open input file |
21:56.49 | nvrpunk | sec |
21:57.08 | hi365_m | yup - wrong hostname! |
21:57.10 | riddlebox | awesome, now sla is working, I wonder if there was just a bug in 1.4.17 |
21:58.53 | nvrpunk | seanbright-home, yeah, unable to open input file |
21:58.55 | nvrpunk | weird |
21:59.00 | seanbright-home | hmmm |
22:00.00 | seanbright-home | nvrpunk: this works for me |
22:00.02 | seanbright-home | for a in /var/lib/asterisk/sounds/*.wav; do b=`echo $a | sed -e 's/ /\\ /g'`; rasterisk -x "file convert $b `echo $b | sed -e s/\.wav$/.g729/g`"; done |
22:00.46 | nvrpunk | hmm |
22:00.55 | nvrpunk | maybe asterisk doesnt do input wav* |
22:01.31 | seanbright-home | i just said it works for me |
22:01.46 | seanbright-home | i just converted all of my wavs to gsm (don't have g729) |
22:02.12 | jameswf-home | http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk |
22:02.23 | nvrpunk | jameswf-home, already read that |
22:02.32 | nvrpunk | i used germanix before |
22:02.41 | nvrpunk | but get a slight glip at the end of every file |
22:02.42 | seanbright-home | nvrpunk: can you pastebin the output you are seeing? |
22:02.46 | seanbright-home | ~pb |
22:02.47 | 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 |
22:03.01 | nvrpunk | Unable to open input file: /root/soundset/sounds/conf-hasleft.wav |
22:03.05 | nvrpunk | for every file |
22:03.09 | nvrpunk | no need to pastebin |
22:03.16 | seanbright-home | asterisk running as root? |
22:03.22 | nvrpunk | seanbright-home, yes |
22:03.39 | nvrpunk | i even chmod 777 the files |
22:03.58 | seanbright-home | wtf? |
22:04.05 | jameswf-home | what was wrong with s'ox |
22:04.14 | nvrpunk | does sox do g729 |
22:04.15 | nvrpunk | i asked |
22:04.33 | nvrpunk | seanbright-home, was trying to see if quality is better using rasterisk and official digium codec |
22:04.55 | seanbright-home | nvrpunk: not based on a quick google, but i could be wrong. |
22:05.03 | *** join/#asterisk philipp64 (n=chatzill@pool-71-112-32-245.sttlwa.dsl-w.verizon.net) |
22:06.11 | seanbright-home | nvrpunk: what version of asterisk? |
22:06.35 | nvrpunk | seanbright-home, its converting the gsm's I have |
22:06.43 | nvrpunk | must be something with the wav sample rate |
22:06.47 | nvrpunk | that asterisk isnt converting |
22:07.04 | seanbright-home | nvrpunk: gotcha. so you're good or no? |
22:07.12 | nvrpunk | good ;) |
22:07.16 | seanbright-home | good. |
22:07.16 | nvrpunk | many thanks |
22:07.20 | nvrpunk | nice script btw |
22:07.23 | seanbright-home | heh |
22:07.40 | seanbright-home | i write all my production code in bash |
22:08.17 | nvrpunk | heh |
22:08.33 | nvrpunk | my boss asked me how to set the proxy to not load a gui |
22:09.52 | jameswf-home | tit tit tit ta tat atta tatt BAM there it is |
22:10.07 | *** join/#asterisk Phillhun1 (n=PHunt@3.72.233.220.exetel.com.au) |
22:12.49 | Phillhun1 | Hi I have a question I have an NEC IPK2 and a sip trunk card that I want to connect to a Cisco call manager but it looks like i need a Sip Proxy or Registrar server to go between the 2 can anyone help |
22:13.06 | nvrpunk | hmm |
22:14.28 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
22:14.33 | nvrpunk | seanbright-home, conversion sounds like crap :/ |
22:14.36 | nvrpunk | compared to the other |
22:14.37 | nvrpunk | haha |
22:14.40 | seanbright-home | heh |
22:14.46 | seanbright-home | well then it was all worth it |
22:15.04 | nvrpunk | germanix did better quality but theres a little noise at the end of each file |
22:16.19 | ManxPower | Phillhun1: Perhaps you should ask on a channel that has to do with the software/hardware you are using. This is not a general VoIP channel. |
22:16.35 | *** join/#asterisk ThatKidKel (n=Kelvin@lacasa.thatkidkel.net) |
22:17.30 | ThatKidKel | is there a way to disable music on hold for a particular call |
22:17.34 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
22:17.34 | *** mode/#asterisk [+o russellb] by ChanServ |
22:17.38 | Phillhun1 | Hi I was hoping that someone would no if Asterisk could be setup as a Sip proxy/Registrar server |
22:17.41 | ManxPower | ThatKidKel: yes. |
22:17.55 | ThatKidKel | ManxPower.. Could you point me to some documentation or give me the command.. |
22:18.03 | ManxPower | Phillhun1: You would use the call manager running SIP as the sip proxy/registrar |
22:18.09 | [TK]D-Fender | ThatKidKel, Set it to a class that has nothing to do. |
22:18.30 | ManxPower | ThatKidKel: you have to enable MoH using the "m" opton to Dial. So no MoH is the default. |
22:18.32 | ThatKidKel | Phillhun1.. Check into Openser.. |
22:19.00 | ThatKidKel | Manx. I have no options on my dial command; and its still giving me music.. |
22:19.14 | ManxPower | ThatKidKel: then you don't live in the same universe as the rest of us. |
22:19.32 | ManxPower | perhaps you don't mean MoH when dialing, but MoH when a caller is on hold? |
22:19.48 | ManxPower | or maybe you mean MoH when a caller is in a queue? |
22:20.07 | ManxPower | I can only guess at what you mean. |
22:20.12 | Phillhun1 | apparently Call manager doesn't do sip proxy |
22:20.28 | ManxPower | Phillhun1: neither does Asterisk, but you don't need a REAL sip proxy. |
22:20.30 | ThatKidKel | ManxPower.. I have no options on my dial command.. And when a call is placed on hold, it starts palying music.. |
22:20.40 | ManxPower | you just need a place to send calls to, Cisco will do that. |
22:20.55 | ManxPower | ThatKidKel: set the musiconholdclasss to something that does not exist |
22:21.49 | ThatKidKel | I'm letting my calls from the PSTN hit Asterisk instead of OpenSER, and I must say, I kinda like it. |
22:22.04 | Phillhun1 | the problem is the NEC will only talk to a sip proxy |
22:25.38 | ManxPower | Phillhun1: the NEC CANNOT know if it is talking to a SIP proxy or a SIP registrar or a SIP B2BUA |
22:25.56 | ManxPower | All it can know is that it sends SIP packets to a destination that doesn't reject them |
22:28.53 | ManxPower | Phillhun1: pretty much all SIP Phones say they need a proxy |
22:31.06 | Phillhun1 | I spoke to an NEC engineer who said they are only able to talk to a Sip Server / Proxy |
22:31.45 | chavigny | yo |
22:33.05 | robmac67 | NEC in the UK use Grandstream ATA's |
22:34.10 | *** join/#asterisk esaym (n=user@72.183.198.134) |
22:44.05 | nvrpunk | seanbright-home, sln to g729 == good gsm to g729 == bad |
22:44.18 | seanbright-home | nvrpunk: well that makes sense |
22:44.26 | seanbright-home | sln is uncompressed |
22:44.32 | nvrpunk | aye |
22:44.33 | nvrpunk | :) |
22:44.41 | Qwell | gsm to g729 == terrible |
22:44.46 | nvrpunk | <-- paid for custom prompts |
22:44.54 | nvrpunk | but they didnt provide g729 |
22:44.56 | jql | the other way sucks too |
22:45.18 | nvrpunk | so, does anyone have any experience with the portech gsm372? |
22:45.30 | nvrpunk | ive spent a couple of hours trying to get inbound to work on it |
22:45.37 | nvrpunk | outbound works great |
22:45.46 | jql | what happens inbound? |
22:45.56 | nvrpunk | it just rings and rings |
22:45.56 | nvrpunk | :) |
22:46.03 | jql | you should answer it |
22:46.03 | nvrpunk | doesnt seem to route anywhere |
22:46.07 | nvrpunk | haha |
22:46.13 | nvrpunk | it doesnt ring any phone |
22:46.23 | jql | not even asterisk? |
22:46.28 | jql | asterisk can answer phones |
22:46.39 | nvrpunk | hmm |
22:47.33 | nvrpunk | <PROTECTED> |
22:47.34 | nvrpunk | <PROTECTED> |
22:47.39 | nvrpunk | thats all i see in asterisk |
22:47.48 | *** part/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
22:48.30 | jql | well, you have a problem there |
22:48.42 | nvrpunk | on the gsm gateway I see the inbound as 0790XXXXXXX@1012 |
22:49.14 | nvrpunk | jql, im passing out one sim out the gateway asiacell, and trying to call back in |
22:49.14 | jql | but the gateway never feels the urge to answer? |
22:49.32 | nvrpunk | jql, the gateway sees the incoming call |
22:49.36 | nvrpunk | just nothing happens with it |
22:49.39 | jql | sees it indeed |
22:50.02 | jameswf-home | alliot of gsm gateways use revpol |
22:50.11 | jameswf-home | allot |
22:50.36 | nvrpunk | http://www.voip-info.org/wiki/view/Setup+MV-370+GSM+Gateway+with+Asterisk |
22:50.41 | nvrpunk | thats the gsm gateway i haave |
22:50.49 | nvrpunk | i used it's context for the incoming |
22:51.18 | nvrpunk | ive also changed it and tried making it just Dial(SIP/sip3,30) a phone i have |
22:52.07 | ManxPower | nvrpunk: that is all you SHOULD expect until the far end answers. |
22:52.19 | ManxPower | setting callprogress=yes would certinally screw things up |
22:52.54 | ManxPower | but I don't think that's the case there, since I don't believe chan_sip supports callprogress=yes (this is fake call progress, not the reall call progress devices provide) |
22:53.32 | nvrpunk | ManxPower, yes but chaging it to just Dial(SIP/sip3) i never get a ring on my phone |
22:53.51 | nvrpunk | which i would imagine the gateway should just link up and dial the phone |
22:54.04 | ManxPower | nvrpunk: there is no difference between 30 and no 30 except for asterisk timing out after 30 seconds |
22:54.13 | nvrpunk | yeah got that |
22:54.18 | nvrpunk | and going to step 2 |
22:54.20 | ManxPower | nvrpunk: yes, it should. contact the gateway's tech support |
22:54.27 | ThatKidKel | if Asterisk is out of the media path--is there anyway to put it back in? |
22:54.35 | nvrpunk | ManxPower, all asian people who prolly wont help :) |
22:54.55 | ManxPower | nvrpunk: I doubt any of us can help either. It sucks to be you. |
22:55.13 | ManxPower | all people here can do is suggest random things that won't work, as the issue is IN THE GATEWAY. |
22:55.44 | jameswf-home | ~random |
22:56.52 | jameswf-home | ~random |
22:57.03 | cesar_CR | hello guys who knows the DMLink cards ??? |
22:59.18 | jameswf-home | ~clones |
22:59.24 | jameswf-home | ~clone |
22:59.25 | jbot | hmm... clone is a clone card - i.e. a worthless, unreliable piece of junk. Is all that extra headache really worth the few dollars you're about to save? |
22:59.54 | jql | but... I get to save the money *now* |
23:00.10 | cesar_CR | ok |
23:00.31 | jameswf-home | well you save all sorts of money if your time is worthless |
23:00.56 | jql | my time is obviously worthless. *points at boss* |
23:02.06 | jameswf-home | If your an hourly employee encourage your boss to go cheap |
23:02.14 | MatBoy | I don't see any verbose messages in the cli when I add this to the logger for cli and start the cli with many v's |
23:02.55 | jql | the cli's -vs matter little |
23:02.55 | *** join/#asterisk comprookie2000 (n=comprook@adsl-065-012-210-216.sip.bct.bellsouth.net) |
23:03.02 | jql | core set verbose 10 or whatever |
23:03.07 | MatBoy | ok |
23:03.11 | riddlebox | man I am so excited now I can tell me boss that I can do sla and we will have a happy customer in a couple weeks when we install their system |
23:03.27 | jameswf-home | you need to properly setup logger.conf |
23:03.45 | MatBoy | jameswf-home, seems to be OK for what I find online |
23:07.59 | ManxPower | riddlebox: have you actually TESTED SLA? |
23:08.56 | riddlebox | ManxPower, I just tested it with 1 line, hopefully next week I can get into the office to test on 4 lines |
23:13.30 | jameswf-home | Unable to register tone zone 'us' means something pooched right? |
23:18.38 | *** join/#asterisk seanbright-home (n=seanbrig@c-69-251-175-43.hsd1.md.comcast.net) |
23:20.02 | tzafrir_home | right |
23:20.32 | tzafrir_home | try the latest zaptel (1.4.9.2) . |
23:20.43 | tzafrir_home | Or just the memset patch to tonezone.c |
23:20.51 | tzafrir_home | jameswf-home, ==^ |
23:22.57 | jameswf-home | known 1.4.8 bug? this is an install from those lime green guys.. |
23:25.12 | drmessano | Mmm Lime Jello |
23:25.48 | jameswf-home | known 1.4.8 bug? this is an install from those lime green guys.. i am like flippin retarded |
23:25.57 | jameswf-home | bah |
23:26.03 | jameswf-home | proves my point |
23:26.22 | jameswf-home | is there a bug id i am like flippin retarded |
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23:40.05 | ManxPower | jameswf-home: if you were on the mailing lists you would know this already. |
23:40.18 | docelmo | Anyone in here have any experience with call files and the CDR's they write? |
23:42.47 | *** join/#asterisk NiklasH_work (n=niklash@triton.dsv.su.se) |
23:43.58 | jameswf-home | heh I get 400 emails a day from the list I look at keywords... |
23:44.19 | NiklasH_work | hi, i hope i'm in the right channel: I have a problem using a phone adapter for outgoing calls via asterisk behind NAT. Inbound calls work fine, but outbound give a busy tone right after the first connect. Anyone have any ideas as to what could be wrong? |
23:44.43 | teknoprep | ~nat |
23:44.44 | jbot | i guess nat is Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
23:45.00 | teknoprep | ~sip_nat.conf |
23:45.12 | NiklasH_work | a software client works fine, so i don't think the nat conf is wrong |
23:45.16 | teknoprep | i don't remember which one it is |
23:45.29 | teknoprep | is it asterisk that is having problems ? |
23:45.33 | teknoprep | or is it your sip client |
23:46.01 | NiklasH_work | yes, just the adapter. softphone works fine. |
23:46.14 | NiklasH_work | i've tried two different adapters, none works |
23:46.18 | teknoprep | is the adapter on the same subnet as your asterisk box ? |
23:46.31 | NiklasH_work | yes, all on the same network |
23:46.33 | dacs | ~astercrm |
23:47.10 | jameswf-home | only bug in my email is loadzone=au says works as loadzone=us |
23:47.33 | NiklasH_work | from the debug info from asterisk, it seems that the adapter places the call ok, then switches to busy after the ok from asterisk |
23:48.35 | ManxPower | jameswf-home: what card do you have again? |
23:48.47 | dacs | is there is a channel for astercrm |
23:49.49 | ManxPower | Re: [asterisk-users] DTMF tone crashes server (Asterisk 1.4.18 with Digium TE120P) |
23:50.29 | ManxPower | NiklasH_work: you must disable all NAT features on the adapter. Asterisk's nat features handles all that stuff |
23:50.55 | NiklasH_work | iĆ've tried doing that, but i could have missed something. iĆ'll check again |
23:51.20 | jameswf-home | no cards in at this point just loaded on vmware |
23:51.31 | jameswf-home | ztdummy |
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23:51.46 | ManxPower | Ah, so it's not the bug I was thinking of. |
23:51.52 | adeel | how do i get * 1.4 to record the cdr's in mysql? |
23:51.56 | adeel | is there an addons package? |
23:51.59 | *** part/#asterisk amir_ (n=amir@unaffiliated/amir) |
23:52.11 | ManxPower | adeel: yes |
23:52.13 | jql | there is |
23:52.20 | jameswf-home | I am just curious if its 1.4.8 or the green people |
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23:53.46 | adeel | ManxPower, ah, thanks |
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23:57.34 | drmessano | Where is host=blah&blah documented? |
23:58.32 | drmessano | Wasn't aware that & was a valid option, and need to know how multiple hosts are handled, in this case, with a peer to an ITSP |
23:59.28 | adeel | drmessano, i always thought you had one host per host= entry, but can have multiple host= entries |
23:59.43 | adeel | e.g., host=foo |
23:59.45 | adeel | host=bar |