| 00:00.21 | Spacy | Okay. So i got another question - if i have a call from chan_capi -> chan_sip, everything works fine. If i have a chan_capi -> voicemail connection, the sound gets chopped. chan_sip -> voicemail also works fine. Any suggestions? (Sorry to be such an annoyance ;) |
| 00:01.08 | hunmonk | hmmhesays: http://pastebin.ca/141451 <-- that's what i have set up so far in iax.conf, and extensions.conf |
| 00:01.23 | hunmonk | hmmhesays: lemme enable debugging now and see what i find |
| 00:02.19 | Skyelar | kavit: I'd say no: it's still open, and there's no message about the patch being committed to either trunk or the 1.2 branch. |
| 00:02.39 | hmmhesays | i wouldn't use _. extension |
| 00:02.50 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 00:03.08 | kavit | Skyelar: so one would have to patch chan_sip.c by hand? |
| 00:03.43 | Skyelar | kavit: yes. You might find it's a bit tricky to do too, as that patch is targetted at the SVN trunk, not the 1.2 branch |
| 00:04.31 | kavit | Skyelar: ah alright thanks for your help |
| 00:06.00 | hunmonk | hmmhesays: yeah, i know. at this point i'm just throwing everything i have at the problem trying to get something to work :) |
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| 00:09.14 | hunmonk | hmmhesays: http://pastebin.ca/141468 <-- results from the iax debug for the attempted call |
| 00:10.10 | hmmhesays | username/password doesn't match |
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| 00:10.56 | hunmonk | hmmhesays: where are you seeing that in the results? |
| 00:12.01 | hunmonk | hmmhesays: also looks to me like 'it's taking the domain as the context?? i'm quite confused as to the proper way to place a call to another iax user... |
| 00:12.27 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 00:12.54 | shodan | how hard is it to unlock a pap2 ? |
| 00:13.02 | wwalker | recommendations for a linux based sip soft phone for testing? (I'll use polycom at the office, but at home, I've just got my server and my notebook, both linux) |
| 00:15.21 | hunmonk | hmmhesays: CAUSE : No authority found <-- is that what indicates the user/pass don't match? |
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| 00:27.30 | quid246 | wwalker: check the wiki, surely something there |
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| 00:35.57 | AndyCap | wwalker: ekiga, x-lite, kphone? |
| 00:36.41 | Kumba_ | What's the command i'd used if I want asterisk to play a file if it's not monday-friday, 9-5? |
| 00:36.51 | Kumba_ | basically, play a "we're closed" file |
| 00:37.30 | *** join/#asterisk RoyK (n=roy@155.80-203-78.nextgentel.com) |
| 00:39.16 | ariel_ | Kumba_, if you look at the /usr/src/asterisk/configs/extensions.conf.sample there is a time ivr there that will help you out. |
| 00:39.33 | Kumba_ | Hmmm... *looks* |
| 00:39.35 | *** join/#asterisk Klitz (n=Klitz@24.96.201.72) |
| 00:41.02 | Kumba_ | hmm... gotoiftime... |
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| 00:48.22 | Kumba_ | If i'm transferring from an incoming context (that begins with s,1,answer) to another context, do I still need to start the second context with answer? |
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| 00:54.21 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 00:56.59 | Kumba_ | hmm... if I was to guess... i'd say you only issue an answer when you want to play something on the line... but not for just general call routing right? |
| 00:58.12 | florz | Kumba_: right - and it affects the channel, not the context, so you only need to do it once per channel, much like picking up the telephone is required only once per call ... |
| 00:58.44 | Kumba_ | ok... so if in my [incoming] context I issue an answer, any sub-contexts wont need it... |
| 00:58.46 | florz | or, more exactly, the "channel instance", as in a call |
| 00:59.04 | Kumba_ | or any contexts issued from the incoming context... |
| 00:59.13 | florz | Kumba_: Well, when the phone is picked up, you can't pick it upper ;-) |
| 00:59.42 | Kumba_ | But would it hurt anything if I had it in there anyways? (incase I use a context somewhere else where it does need to pickup?) |
| 01:00.50 | florz | Kumba_: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Answer =:-) |
| 01:01.35 | *** join/#asterisk dos000 (n=dos000@i216-58-93-204.cybersurf.com) |
| 01:01.37 | Kumba_ | exten => i,1,sendnuke() |
| 01:02.26 | florz | Well, I'd probably do that using AGI ... well, then again, I'd probably rather not do it at all ;-) |
| 01:03.15 | Kumba_ | This dialplan stuff makes my head hurt... and i'm only doing 2-lines and 6-ext's... (soho) |
| 01:04.07 | *** part/#asterisk hunmonk (n=hunmonk@pool-71-97-41-106.dfw.dsl-w.verizon.net) |
| 01:04.27 | Kumba_ | florz: You dont know if asterisk has the ability to send SMS messages to a cell phone do you? |
| 01:04.50 | Kumba_ | I guess I could make the emergency-call-routing record someone, then call someone's cell phone and play it back to 'em... |
| 01:05.58 | rg1_ | I need to send an AGI argument that includes a line-feed from a dial-plan - can anyone help me know how to "encode" that? |
| 01:06.36 | quid246 | kumba: I'm sure I've seen an SMS app omewhere for *, but don't quote me |
| 01:07.28 | rg1_ | kumba - you can do that with a multi-tech gsm modem |
| 01:07.49 | rg1_ | not necessarily from asterisk, but if you need to send it from asterisk you can do it with an AGI script |
| 01:07.50 | florz | Kumba_: Well, there is some modem stuff and an SMS-or-so application that is capable of communicating with SMSC or something - but never used that, so no real clue ... |
| 01:08.09 | *** join/#asterisk anthonyu (i=user347@netblock-68-183-123-247.dslextreme.com) |
| 01:08.39 | florz | Kumba_: Otherwise, as rg1_ says, or you could also use some analog phone modem for calling up the SMSC |
| 01:08.40 | rg1_ | hey - can someone help me with the insertion of a line-feed in a "string" from a dial-plan |
| 01:09.05 | florz | rg1_: No clue, but why do you need that? |
| 01:09.13 | rg1_ | florz - no its not an analog modem - you actually put a gsm chip in the modem |
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| 01:09.42 | Kumba_ | that sounds like way too much work... |
| 01:09.43 | rg1_ | florz - i need to send lines of data so that a TTS app I have will see a line feed and give me a more extended pause |
| 01:09.46 | Kumba_ | I'll just tell them no :) |
| 01:09.53 | florz | rg1_: That's why I said "or you cold [...]" :-) |
| 01:10.25 | florz | erm, s/cold/could/ |
| 01:10.34 | rg1_ | so I put like "Hello world. [...] How are you" |
| 01:10.39 | rg1_ | is that what you are saying |
| 01:10.50 | rg1_ | and the [...] will give me a long pause? |
| 01:11.09 | florz | rg1_: gnah! |
| 01:11.15 | rg1_ | ? |
| 01:11.19 | rg1_ | HELP ME :) |
| 01:11.26 | Kumba_ | I think that the context that [...] was passed to from came from the [rg1-answer_to_kumba's_question] context... |
| 01:11.26 | anthonyu | hi, i want to send callers to various extensions based on caller-id and number called. should i do that with agi, ami or something simpler? |
| 01:11.28 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 01:11.41 | Kumba_ | or something like that... |
| 01:11.42 | florz | rg1_: You really seem to misunderstand basically everyhing one could not possibly misunderstand =:-) |
| 01:11.44 | shodan | are grandstreams handytone 386 good with * ? I'm thinking of getting one from voipsupply with 1 or 2 voip phones (not decided which yet) |
| 01:11.51 | rg1_ | florz, so true |
| 01:12.01 | Kumba_ | I need more beer to understand this extensions.conf stuff |
| 01:12.14 | rg1_ | so if i want to get a long pause in a TTS string, can you please tell me how to do that? |
| 01:12.18 | Kumba_ | Screw mountain dew... give me some Stella Artois... |
| 01:12.52 | florz | rg1_: Well, how about using some random string you make up and do the conversion in the AGI? |
| 01:13.25 | rg1_ | didn't think of that |
| 01:13.32 | rg1_ | ok, i will give that whirl |
| 01:13.45 | rg1_ | try my multi-tech gsm modem thing - it worked |
| 01:13.50 | rg1_ | adios |
| 01:14.46 | Kumba_ | rg1: thanks for the input... I appreciate it... |
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| 01:30.44 | shodan | is this good for use with * => http://www.voipsupply.com/product_info.php?products_id=518 ? |
| 01:38.25 | *** part/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 01:44.07 | bkw_ | rg1_, they have a markup language for TTS to insert a pause |
| 01:44.21 | bkw_ | rg1_, if you were to look at mod_rss in freeswitch you can see how we did it with Cepstral |
| 01:44.23 | wwalker | I've read lots of articles talking about Asterisk on a WRT54G. Has anyone actually done it? any idea how many real calls it can handle?? Assume no transcoding, I would force the phones to use the same codec as the VOIP provider |
| 01:44.26 | *** join/#asterisk uk-wombat (n=root@82.163.6.212) |
| 01:44.35 | bkw_ | and If things go right i'll have mod_festival done this next week followed by mod_sphinx |
| 01:53.01 | doolph | where can I get certified online |
| 01:53.13 | nevyn | in what? |
| 01:53.16 | DrRighteous | wwalker: asterisk on wrt54g will work, but I wouldn't try voicemail (lack of storage, unless you hw upgrd), and # of channels of about 24. |
| 01:53.23 | doolph | asterisk/digium |
| 01:53.31 | bkw_ | why on earth would you wanna get certified? |
| 01:53.49 | puzzled | bkw_: maybe the BofH likes it |
| 01:53.51 | DrRighteous | I would suggest SER on wrt54g, which will allow for inside lan phone 2 phone communication, and remotely hosting the asterisk/pbx box |
| 01:53.51 | doolph | because a company is asking me for a certification |
| 01:53.52 | techie | ywah |
| 01:53.57 | bkw_ | they need to get a clue |
| 01:53.59 | nevyn | DrRighteous: there's always the versions with usb on them put a 512mb flash drive on it and you're laughing |
| 01:54.01 | bkw_ | the cert means NOTHING |
| 01:54.09 | nevyn | bkw_: they generally don't |
| 01:54.24 | bkw_ | Those who can do.. will do.. those that can't, get certified! |
| 01:54.39 | doolph | or I just make my own certification |
| 01:54.44 | doolph | rofl |
| 01:54.50 | bkw_ | I am considered the Anti-Asterisk these days! |
| 01:54.55 | DrRighteous | nevyn: a wrt54g with usb? |
| 01:55.01 | Nivex | bkw_: and yet here you are |
| 01:55.06 | nevyn | DrRighteous: well wrt54g clone |
| 01:55.13 | techie | by whom |
| 01:55.16 | nevyn | the asus WL500G springs to mind |
| 01:55.30 | bkw_ | Nivex, yep. I still have to support asterisk installations for a year or so more. |
| 01:55.59 | nevyn | bkw_: what's your preference? |
| 01:56.10 | wwalker | DrRighteous! Thanks! I'm using a WGT634U actually so have local USB storage of a hlaf a Gig. But 24 channels is way more than I thought it would handle! |
| 01:56.12 | Kumba_ | Does this macro make sense? http://pastebin.ca/141656 |
| 01:56.23 | Kumba_ | Basically, I call it, and pass the extension, and it routes it... |
| 01:56.29 | bkw_ | nevyn, well since asterisk isn't carrier grade we had to write something new from scratch. You might have heard of it.. its called FreeSWITCH.... http://www.freeswitch.org |
| 01:56.30 | *** join/#asterisk oej (n=oej@63.116.149.163) |
| 01:56.39 | nevyn | yeah I've heard of it |
| 01:56.43 | nevyn | gpl? |
| 01:56.45 | bkw_ | MPL |
| 01:56.51 | nevyn | fair enough |
| 01:57.14 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 01:57.31 | bkw_ | so far we can't find a sip stack that will take 100cps at 100ms call duration for longer than a few min before the sip stack takes a shit and dies. |
| 01:57.41 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 01:57.57 | bkw_ | Nobody out there makes a good sip stack. |
| 01:58.19 | bkw_ | even Asterisks sip stack will die quickly under the same test conditions |
| 01:58.28 | puzzled | bkw_: did you try the Nokia one too (or was that ericsson)? |
| 01:58.36 | bkw_ | that one is even worse |
| 01:58.54 | bkw_ | the requirements we have for software are much higher than most Open Source software can meet. |
| 01:59.10 | bkw_ | 1. Cross Platform supporting at the very least Windows, Linux and Mac OS X. |
| 01:59.21 | bkw_ | 2. Willing to accept patches for bugs we find or features we add. |
| 01:59.36 | bkw_ | we have a few projects that have been helping really well. |
| 01:59.39 | *** join/#asterisk TheCops (i=nobody@got.securebinary.com) |
| 01:59.55 | nevyn | bkw_: so your stack does that? |
| 01:59.56 | bkw_ | but in all honesty Open Source projects have to work together to accomplish goals... |
| 02:00.13 | bkw_ | nevyn, we don't have a stack that is stable yet. We are looking at the OpenSolaris SIP Stack. |
| 02:00.14 | nevyn | bkw_: hericy |
| 02:00.23 | nevyn | working together... |
| 02:00.29 | nevyn | this is MY project goddamnit |
| 02:00.37 | nevyn | kde rocks gnome sucks !! etc |
| 02:00.45 | bkw_ | we do work with all the projects we depend on. |
| 02:01.03 | bkw_ | and so far they have all been good to work with us in both directions. |
| 02:01.09 | JT | does SER stack up? |
| 02:01.23 | bkw_ | The SER stack isn't something you can break apart |
| 02:01.24 | wwalker | nevyn: if you keep talking that way the BSD guys wills tart fighting over whose side of theflame war you're going to be on. |
| 02:01.28 | bkw_ | its GPL and not compatible with MPL |
| 02:01.35 | JT | right |
| 02:01.43 | JT | but does it handle the load you were speaking of? |
| 02:01.54 | bkw_ | I think it can. |
| 02:01.54 | *** join/#asterisk rushowr (n=rushowr@cpe-24-26-133-106.columbus.res.rr.com) |
| 02:01.56 | benjk | bkw_ w've been there before |
| 02:02.05 | benjk | nobody made a good OSI stack either |
| 02:02.05 | rushowr | hey gentlemen |
| 02:02.23 | rushowr | anyone know why I'd get the message "no hardware transcoders found" (zaptel card) |
| 02:02.23 | nevyn | what's the use of a 100ms call? |
| 02:02.28 | benjk | white elephants are never materialising in form of good implementations |
| 02:02.32 | bkw_ | but I doubt the SER stack is reentrant |
| 02:02.47 | bkw_ | nevyn, because thats when you have race conditions in call setup and tear down |
| 02:02.51 | bkw_ | once the call is up you're good |
| 02:02.59 | bkw_ | but if you do lots of short calls you'll expose race conditions faster |
| 02:03.04 | nevyn | ah |
| 02:03.07 | bkw_ | its the true test |
| 02:03.17 | rushowr | (sorry to repost, but a client is on my ass) anyone know why I'd get the message "no hardware transcoders found" (zaptel card) |
| 02:03.36 | bkw_ | rushowr, thats a new one to me. |
| 02:03.37 | anthonyu | how do i send people who are in a list of caller IDs directly to voicemail? |
| 02:03.44 | rushowr | I get the message on startup of asterisk (latest trunk) |
| 02:03.53 | bkw_ | exten => 555/918555121,1,Voicemail |
| 02:03.54 | nevyn | anthonyu: ooh I want to know that one.. |
| 02:04.10 | file | rushowr: you don't have a TC400P card, so therefore no hardware transcoder |
| 02:04.11 | nevyn | also how good/flexible is the rouuting code? |
| 02:04.13 | bkw_ | its exten => XXX/CIDHERE,1, |
| 02:04.14 | anthonyu | nevyn, is that like something to do in AGI or AMI? |
| 02:04.16 | rushowr | hrm.... |
| 02:04.20 | bkw_ | nevyn, for? |
| 02:04.25 | benjk | anthonyu, you create a dictionary in astdb |
| 02:04.33 | rushowr | file, so shouldn't cause issues with dialing out over the card? |
| 02:04.34 | nevyn | bkw_: routing based on Caller ID |
| 02:04.36 | rushowr | I do have a card |
| 02:04.44 | bkw_ | nevyn, in asterisk or freeswitch? |
| 02:04.47 | nevyn | either. |
| 02:04.48 | file | rushowr: no, your problem is elsewhere |
| 02:04.52 | rushowr | ok thx |
| 02:04.59 | nevyn | bkw_: /both |
| 02:04.59 | benjk | database foobar nnnnnnnn 1 |
| 02:05.00 | anthonyu | benjk, if i look up astdb and dictionary on google, i can figure it out? |
| 02:05.01 | JT | wouldn't that entirely depend on how you implemented your callerid routing system? |
| 02:05.04 | bkw_ | well asterisk is hard coded on the dial plan format. Freeswitch is 100% plugable on the dialplan options. |
| 02:05.04 | *** part/#asterisk rushowr (n=rushowr@cpe-24-26-133-106.columbus.res.rr.com) |
| 02:05.09 | nevyn | hrm |
| 02:05.10 | benjk | sorry |
| 02:05.11 | nevyn | that's neat |
| 02:05.19 | benjk | database put foobar nnnnnnnnn 1 |
| 02:05.28 | nevyn | bkw_: so if I wanted to get fancy |
| 02:05.32 | *** join/#asterisk xachen (i=justin@pdpc/supporter/student/xachen) |
| 02:05.40 | benjk | for every number |
| 02:05.47 | benjk | where nnnnnnnnn is the number |
| 02:05.58 | nevyn | route on callerid to your representative is the plan |
| 02:06.15 | nevyn | looking up your cid number in the database to get the extension of your agent |
| 02:06.16 | bkw_ | nevyn, http://www.freeswitch.org/docs/structswitch__loadable__module__interface.html |
| 02:06.22 | anthonyu | ah |
| 02:06.22 | benjk | then in your incoming context you test the callerID against the foobar dictionaty (family in astdb lingo) |
| 02:06.25 | TheCops | There's way to specify a range of RTP port PER sip device ?! |
| 02:06.36 | benjk | if it is present you goto voicemail, if not you continue |
| 02:06.41 | bkw_ | nevyn, yes you can even treat calls for X or Y module different ... because you get the name of the module doing the lookup |
| 02:06.50 | anthonyu | thank you |
| 02:07.21 | bkw_ | nevyn, also if you want to add API calls you don't have to hack the core to do so... you just create a module to extend the API |
| 02:07.27 | nevyn | col |
| 02:07.53 | nevyn | are dictionarys going to do what I want? |
| 02:08.06 | nevyn | bkw_: is there a featurelist for freeswitch somewhere? |
| 02:08.14 | bkw_ | lets talk in private |
| 02:12.00 | Kumba_ | _.X = Match Everything right? |
| 02:12.18 | xachen | _. |
| 02:12.39 | Kumba_ | danke |
| 02:12.40 | bkw_ | but since people dont know how to use _. it gives you a "moron" warning if you use _. |
| 02:12.48 | *** join/#asterisk ManxPower (n=eric@69-2-85-41.wan.networktel.net) |
| 02:12.56 | Kumba_ | Yes, I am definitely a moron at dialplans :D |
| 02:12.58 | Kumba_ | so that's ok |
| 02:13.26 | doolph | rofl |
| 02:13.36 | SwK_ | carefule w/ _. as it matches everything (including special extens like h) |
| 02:13.43 | bkw_ | only |
| 02:13.49 | bkw_ | if its directly used in a context |
| 02:13.56 | bkw_ | you always use it in its own context then include it |
| 02:14.00 | bkw_ | so its considered secondary |
| 02:14.00 | SwK_ | yeah |
| 02:14.06 | bkw_ | otherwise you mess up |
| 02:14.08 | SwK_ | but you know people wont do that ;) |
| 02:14.20 | bkw_ | you don't even wanna get me started |
| 02:14.23 | Kumba_ | So what you all are really saying is to just bite the bullet and set up all the dialing patterns... |
| 02:14.27 | SwK_ | heh |
| 02:15.22 | *** join/#asterisk dos000 (n=dos000@i216-58-92-66.cybersurf.com) |
| 02:20.24 | ManxPower | _. is almost never REQUIRED. |
| 02:21.43 | doolph | i have _. |
| 02:21.58 | doolph | in a cc billing |
| 02:22.05 | doolph | that pass the thing to agi |
| 02:22.24 | bkw_ | agi? eww |
| 02:22.36 | bkw_ | agi is good for small tasks where you get in and exit |
| 02:22.40 | bkw_ | but never do a dial from inside an agi |
| 02:22.48 | bkw_ | you're askin for it if you do |
| 02:23.13 | doolph | let me see what it is |
| 02:24.21 | doolph | ah no sorry, its not _. heh they just changed it to the right format |
| 02:25.27 | hmmhesays | yeah |
| 02:25.28 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:25.37 | hmmhesays | I wrote a whole calling card dp without agi |
| 02:25.48 | doolph | cool for you |
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| 02:25.59 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 02:26.01 | hmmhesays | it is pretty nice |
| 02:26.24 | JT | macros, or apps? |
| 02:26.48 | hmmhesays | no macros |
| 02:26.55 | hmmhesays | pretty heavily dependent on cmd mysql |
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| 02:30.47 | SwK_ | that has to be painful |
| 02:31.17 | hmmhesays | not really |
| 02:31.48 | hmmhesays | works pretty well |
| 02:33.51 | suma | will iaxy works with fxo port ? |
| 02:34.09 | SwK_ | and another example of why extensions.conf is less of a config file and more of a scripting language |
| 02:34.18 | Kumba_ | What are the toll-free prefixes? 800/877/888? |
| 02:34.24 | SwK_ | and 866 |
| 02:34.27 | Kumba_ | k |
| 02:34.52 | hmmhesays | more or less yeah |
| 02:36.34 | Kumba_ | Hmmm... anyone got a patter for international calls? I cant seem to find a clear example on voip-info |
| 02:36.47 | hmmhesays | what a fantastically vague question |
| 02:36.54 | *** part/#asterisk DrRighteous (n=DrRighte@ool-457843d1.dyn.optonline.net) |
| 02:36.56 | Kumba_ | rr pattern |
| 02:37.06 | Kumba_ | damn keyboard... needs batteries... |
| 02:38.17 | Kumba_ | well, like local is _NXXXXXX, and domestic is _NXXNXXXXXX... what would international dialing be? |
| 02:39.55 | Kumba_ | _011. ? |
| 02:42.11 | hmmhesays | ban now I can't get my iax client to register |
| 02:43.29 | *** join/#asterisk hunmonk (n=hunmonk@pool-71-97-41-106.dfw.dsl-w.verizon.net) |
| 02:45.59 | hunmonk | hmmhesays: got time for a quick question? |
| 02:46.07 | hmmhesays | i suppose |
| 02:46.57 | *** join/#asterisk W9SH (n=W9SH@adsl-068-209-117-205.sip.asm.bellsouth.net) |
| 02:47.11 | hunmonk | hmmhesays: http://pastebin.ca/141720 <-- getting errors like this when i try to use any other iax softphone. i believe both are using GSM as a codec. any ideas? |
| 02:47.31 | hmmhesays | with the 1000 & 2000 peers? |
| 02:47.38 | hunmonk | yep |
| 02:47.43 | hmmhesays | allow=gsm |
| 02:47.50 | hunmonk | ah, great. thanks! |
| 02:48.23 | hmmhesays | np |
| 02:51.11 | hmmhesays | this is odd |
| 02:54.25 | kavit | has anyone else encountered this bug -> http://bugs.digium.com/view.php?id=7403&nbn=5 |
| 02:56.14 | hunmonk | hmmhesays: ha. the shareware iax phone sounds like ass, the freeware one sounds perfect... :) |
| 02:56.18 | bkw_ | OMG you're joshing me right? Asterisk has bugs? SINCE WHEN? |
| 02:56.35 | hmmhesays | which ones? |
| 02:56.40 | bkw_ | their's more than one? |
| 02:56.47 | suma | bkw_: possible to replace iaxy with fxo module ? |
| 02:56.52 | bkw_ | suma, No |
| 02:56.59 | JT | shareware/freeware... aren't both these terms not quite applicable to open source software? |
| 02:57.17 | Qwell | JT: no, neither are the same |
| 02:57.17 | bkw_ | well why on earth would you even have an IAXy in the first place? |
| 02:57.30 | suma | bkw_: removed the fxs module and used in my tdm400p card ! works fine |
| 02:57.31 | hunmonk | hmmhesays: LoudHush (shareware), JackenIAX (freeware) |
| 02:57.33 | bkw_ | you can buy something thats got more codecs and can do what you want for half the price |
| 02:57.51 | bkw_ | suma, the firmware doesn't deal with FXO i'm pretty sure of that |
| 02:58.46 | suma | bkw_: so, if we change the firmwae we can add fxo module? hardware is ok to work with that? |
| 02:59.01 | bkw_ | doubt digium will even release firmware that would make it an FXO |
| 02:59.07 | bkw_ | go buy a sipura |
| 02:59.32 | bkw_ | SPA-3000 is it? ya ya thats it |
| 02:59.44 | suma | yes |
| 02:59.51 | bkw_ | I have one.. works great |
| 03:00.06 | suma | I thought iax is good compared to sip |
| 03:00.11 | bkw_ | pfft |
| 03:00.24 | bkw_ | I hate the IAX protocol with a passion |
| 03:00.26 | *** part/#asterisk hunmonk (n=hunmonk@pool-71-97-41-106.dfw.dsl-w.verizon.net) |
| 03:00.51 | hmmhesays | I think you just have a special dislike for asterisk as a whole |
| 03:00.51 | bkw_ | SIP and RTP are perfectly capable of busting NAT |
| 03:01.02 | bkw_ | no asterisk has its place |
| 03:01.08 | bkw_ | and I still support it. |
| 03:01.21 | suma | bkw_: that is great |
| 03:01.23 | bkw_ | but we have already proven we can bust tripple NAT with RTP+ICE |
| 03:01.42 | bkw_ | at linuxtag we had calls going thru their tripple nat from here to there using the Jingle protocol with rtp/ice |
| 03:02.26 | suma | bkw_: i c |
| 03:03.08 | suma | bkw_: i had problems with SIPURA in configuring behind NAT |
| 03:03.56 | ManxPower | bkw_, *nod* NAT is NOT a big issue |
| 03:04.10 | suma | bkw_: Asterisk -> NAT -> internet -> NAT (SIPURA) |
| 03:04.58 | suma | did not went quite well through |
| 03:06.33 | eliXier | you need a STUN or VPN, i think |
| 03:06.45 | ManxPower | bkw_, I've been building a CATV system at the campground 8-) |
| 03:07.48 | hmmhesays | not anymore it is not |
| 03:07.52 | *** join/#asterisk NoRemorse (n=bah@eth2462.vic.adsl.internode.on.net) |
| 03:07.56 | NoRemorse | hi all |
| 03:08.27 | *** join/#asterisk JohnJacob (n=m00p@pool-71-127-86-105.aubnin.fios.verizon.net) |
| 03:08.29 | NoRemorse | anyone got any idea what can cause this error please? it happens when trying to dial into asterisk as a guest WARNING[10108]: chan_sip.c:3511 process_sdp: Insufficient information for SDP (m = '', c = '') |
| 03:09.52 | doolph | are you trying to make video |
| 03:10.30 | bkw_ | ManxPower, kewl |
| 03:10.37 | hmmhesays | ok now my iax clients aren't registereing |
| 03:10.43 | bkw_ | I have been looking at doing IPTV using freeswitch as the core |
| 03:12.22 | doolph | anyone know about hotel+asterisk softwares |
| 03:12.27 | hmmhesays | bah binding to the wrong nic |
| 03:14.26 | suma | NoRemorse: you have the SIP Messages belonging to that warning ? |
| 03:15.27 | hmmhesays | doolph: what do you mean? |
| 03:15.48 | doolph | i wasnt talking with you |
| 03:16.04 | doolph | but what you want todo |
| 03:16.38 | JT | doolph: how can anyone know who you're talking to if you don't direct your questions? |
| 03:16.40 | hmmhesays | "anyone know about hotel+asterisk softwares" |
| 03:16.56 | doolph | ah |
| 03:17.20 | doolph | hotel billing with pbx integrated using asterisk |
| 03:17.35 | hmmhesays | what do you want to know? |
| 03:17.53 | doolph | anyone that has or trying to do something like that |
| 03:18.02 | hmmhesays | what do you want to know |
| 03:18.24 | doolph | just that |
| 03:18.37 | doolph | what are the features, etc |
| 03:18.41 | hmmhesays | you going to have to ask a more specific question |
| 03:20.36 | *** join/#asterisk trelane (i=trelane@unaffiliated/trelane) |
| 03:21.17 | bkw_ | translation for what doolph wants is "I'm wanting to just take the work of someone else if they have already done the work" |
| 03:21.34 | lowlevel | wow that was one spicey burrito |
| 03:21.36 | bkw_ | no need to duplicate work |
| 03:22.46 | shodan | how does call waiting fit in with * ? |
| 03:23.48 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:26.12 | Kumba_ | Anyone feel like scanning over a dialplan real quick and telling me if you see any blatantly obvious/newbish/retarded errors? |
| 03:27.07 | CunningPike | Kumba_: pastebin it |
| 03:27.09 | CunningPike | ~pb |
| 03:27.14 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca/, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com/, or http://bzflag.pastebin.ca/, or for images use http://imageshack.us/, or http://www.paste-it.net/ |
| 03:27.14 | Kumba_ | http://pastebin.ca/141784 |
| 03:27.20 | CunningPike | Wow - you're fast ;) |
| 03:27.34 | Kumba_ | Already had it paste bin'd... I just didn't want to incur anyone's wrath by pasting it... |
| 03:27.35 | NoRemorse | suma: yes |
| 03:27.37 | NoRemorse | SIP/2.0 488 Not acceptable here |
| 03:28.56 | CunningPike | Kumba_: ;) This looks invalid to me: Goto(fbd/fbd-closed,s,1) |
| 03:29.22 | CunningPike | Kumba_: Otherwise, nothing else leapt out at me. Are you having a specific problem? |
| 03:29.35 | Kumba_ | no... just first dialplan i've done... |
| 03:29.42 | Kumba_ | not looking to have any specific problems :) |
| 03:30.08 | CunningPike | Kumba_: Best way to proceed is to traverse its logic and make note of any errors |
| 03:30.22 | CunningPike | Kumba_: By calling each extension etc |
| 03:30.36 | Kumba_ | Yeah... i'm just not sure i'm using the proper syntax's on some of the stuff... |
| 03:30.55 | CunningPike | shodan: How are you connected to the PSTN? |
| 03:32.49 | shodan | CunningPike, 2 fxos to 2 lines |
| 03:33.37 | shodan | asterisk in the box with the fxos and soft/hardphones on the lan |
| 03:34.17 | CunningPike | Kumba_: Best thing is to just try it :) |
| 03:34.24 | Kumba_ | CunningPike: Can I use the n directive in a 101, 102, 103 priority? like 'exten => s,10n,hangup' |
| 03:35.02 | CunningPike | Kumba_: I don't believe so - why do you need to? |
| 03:35.18 | Kumba_ | Just curious... |
| 03:35.30 | Kumba_ | had a need where I could have used it... for 3 lines... |
| 03:35.32 | CunningPike | shodan: Hmm - not familiar with how call waiting works on POTS lines |
| 03:36.09 | CunningPike | Kumba_: 'Splain? |
| 03:37.35 | *** join/#asterisk tengulre (n=tengulre@61.185.224.66) |
| 03:38.14 | shodan | CunningPike, you are talking on the phone with 1st caller , then 2nd caller calls , your line does a muffled *beep* *beep* at 3 seconds intervals as long as the 2nd caller is ringing , you flash the line (hangup/pickup quickly) and you switch line to the 2nd caller , after that you can flash the line as often as you want to switch between the callers |
| 03:38.17 | Kumba_ | I had a part where if the dial'd ext timed out, to start with the 101 priority, and I needed to add a line for a change... |
| 03:39.35 | CunningPike | shodan: Yes - what I mean was I'm not sure how it works with SIP phones -> Asterisk -> FXO -> PSTN |
| 03:40.28 | shodan | oh ok |
| 03:41.38 | shodan | I have a cheapo netweb 301 , but it doesn't have a flash button and I'm pretty sure that if I hangup even quickly it will kill the connection (but can't test because call waiting is currently disabled and the phone isn't properly configured yet (can't find the damn cd :( )) |
| 03:42.11 | CunningPike | shodan: Ya - I doubt if flashing works on a SIP phone - our Polycoms would hang up right away |
| 03:42.46 | CunningPike | shodan: There might be a *nn code you can use or something - have you checked the wiki? |
| 03:43.32 | shodan | no, you mean voip-info.org ? |
| 03:46.41 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 03:51.07 | CunningPike | shodan: Yes |
| 03:51.10 | CunningPike | ~thewiki |
| 03:51.11 | jbot | thewiki is probably at http://www.voip-info.org/wiki-Asterisk |
| 03:52.48 | shodan | ah k , I wasn't sure if that was -the- wiki |
| 03:53.04 | shodan | I just checked looks like it's implemented but the phone must have support for it |
| 03:55.56 | CunningPike | shodan: I wonder if there is some dialplan trick....... |
| 03:56.59 | shodan | dunno , I just got * working for the 1st time , my dialplan isn't even done yet |
| 03:57.01 | *** join/#asterisk Kasimeng (i=WinNT@125.215.196.251) |
| 03:57.09 | Kasimeng | Hi Everyone |
| 03:57.32 | Kasimeng | Have any one try Grandstream HT486 with Asterisk? |
| 03:58.55 | shodan | while you're asking I was wondering about getting a 386 so I can use old , more egonomic/less flashy analog phone with my *box |
| 04:00.18 | hmmhesays | ok got my user auth stuff one |
| 04:00.22 | hmmhesays | *done |
| 04:09.47 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 04:09.57 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 04:20.15 | *** part/#asterisk rnovotny22 (n=rnovonty@71-37-225-46.mpls.qwest.net) |
| 04:24.04 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:45.05 | shodan | what ATA should I get to work with * ? (2 fxs under about 100$usd) |
| 04:45.36 | shodan | dlink, granstream or linksys ? |
| 04:49.12 | ManxPower | ~docs |
| 04:49.13 | jbot | [docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 04:49.20 | *** join/#asterisk techie (n=gus@adsl-068-209-242-072.sip.mia.bellsouth.net) |
| 04:49.29 | ManxPower | shodan, SIPura |
| 04:52.48 | shodan | isn't that the same as linksys in a different casing ? |
| 04:53.36 | shodan | http://www.voipsupply.com/product_info.php?products_id=713 ? |
| 04:56.05 | *** join/#asterisk Jenocin (i=jenocin@99.3.118.70.cfl.res.rr.com) |
| 04:56.09 | Jenocin | anyone use hudlite? |
| 05:01.25 | *** join/#asterisk tlow (n=tlow@bgp.terrorist.net) |
| 05:03.51 | *** join/#asterisk mcnobody (n=laaksola@80.95.135.45) |
| 05:07.32 | shodan | ManxPower, should I go with that linksys sipura spa-2002 or get an older version sipura spa-2002 ? |
| 05:14.43 | *** part/#asterisk tlow (n=tlow@bgp.terrorist.net) |
| 05:19.31 | Kasimeng | Does anyone know how to fax with Grandstream and *? |
| 05:26.18 | png6 | if I want to reach a sertain extension when I dial in, and only have one number - is the only way a menu system that the user meets when he first call? |
| 05:26.21 | png6 | or can I dial 012345 and then press the extensionnumber (where 012345 is my phone number) |
| 05:29.24 | *** join/#asterisk somegeek_ (i=levin@tor/regular/somegeek) |
| 05:33.20 | CunningPike | shodan: The Linksys is the same thing. Cisco owns Linksys owns Sipura |
| 05:33.43 | CunningPike | Jenocin: I could never get it working. We're looking at FOP instead |
| 05:35.39 | CunningPike | png6: Both are the same from a dialplan perspective - the only difference is whether you play a sound file or not |
| 05:36.06 | CunningPike | Kasimeng: Grandstream what? ATA? |
| 05:37.16 | shodan | is this possible => I want to get one of those grandstream videophones , can I stream a file (xvid or mpeg2 through mencoder to the proper format maybe) on to the screen ? |
| 05:38.41 | *** join/#asterisk AJaymn (n=boiwonde@70.59.126.206) |
| 05:38.44 | shodan | that phone http://blog.tmcnet.com/blog/tom-keating/voip/grandstream-gxv3000-video-phone.asp |
| 05:39.16 | CunningPike | shodan: I'm sure I haven't the faintest idea :) |
| 05:41.53 | shodan | I heard if you do a voicemail it will record the video , maybe I can transcode my video into files that * uses for voicemail .. |
| 06:01.29 | *** join/#asterisk DarKnesS_WolF (n=wolf@212.103.170.135) |
| 06:04.00 | ManxPower | png6, unless you have some weird European ISDN, you have to dial from a menu |
| 06:05.07 | ManxPower | png6, and the "weird european isdn" setup would still require multiple numbers from the telco |
| 06:07.14 | benjk | what's "weird European ISDN" ? |
| 06:08.16 | JT | maybe he's refering to the type of BRI that most of the planet uses |
| 06:08.17 | ManxPower | benjk, Lets say the standard number length for your country is 10 digits. You can get 12 digit numberd from the telco |
| 06:09.58 | *** join/#asterisk adelas (n=booger@rrcs-24-199-21-138.west.biz.rr.com) |
| 06:10.10 | *** join/#asterisk daysmen3 (n=primus@host86-138-239-164.range86-138.btcentralplus.com) |
| 06:11.33 | JT | anyway, what png6 needs is either a DID for each extension if it must be able to be directly dialled from the PSTN |
| 06:11.48 | JT | or it has to be entered when connected to his asterisk box |
| 06:12.37 | *** join/#asterisk intralanman (n=lanman@pool-72-82-74-171.nrflva.east.verizon.net) |
| 06:19.46 | benjk | I see, I thought you were talking about the protocol itself |
| 06:20.02 | JT | so did i |
| 06:20.05 | JT | weird reference |
| 06:20.15 | JT | "weird european isdn" ;) |
| 06:20.35 | benjk | but I can see the point |
| 06:20.58 | benjk | from a US citizen's point of view most if not all things non-US are "weird" or "bizarre" |
| 06:21.39 | benjk | they even have national league sports that call themselves "world series" and things like that |
| 06:23.22 | JT | haha |
| 06:23.26 | JT | true true |
| 06:23.38 | benjk | we're all extra-terrestrials :) |
| 06:24.10 | JT | is it the same national league that uses metric fucktonnes of padding on their players for a contact sport? |
| 06:24.24 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:24.24 | JT | or should that be imperial fucktons |
| 06:24.34 | benjk | I wouldn't know |
| 06:25.06 | JT | heh |
| 06:25.06 | benjk | but the term imperial for US measurements and stuff seems more and more appropriate |
| 06:25.27 | benjk | switchtype = imperialisdn |
| 06:25.52 | JT | switchtype = proprietaryus |
| 06:25.53 | JT | ;) |
| 06:25.57 | benjk | aka I2 |
| 06:26.41 | benjk | language = imperialusenglish |
| 06:27.04 | JT | heh |
| 06:27.14 | benjk | codec = imperallaw |
| 06:27.25 | benjk | imperiallaw |
| 06:27.44 | benjk | mind you, we use that here in post-imperial Japan |
| 06:28.18 | JT | annoys me when people outside of north america or others using the same telephony standards use G.711u because they don't know any better |
| 06:28.37 | JT | and most of the documentation does not clearly state that most of the world uses a-law |
| 06:28.41 | benjk | nah, NTT uses T1 |
| 06:28.43 | *** join/#asterisk Grnd-Wire (i=GrndWire@67-40-17-231.tukw.qwest.net) |
| 06:28.47 | Grnd-Wire | Good evening gentlemen! |
| 06:28.55 | JT | like the asterisk book |
| 06:28.55 | Grnd-Wire | err - and any ladies that might be in here as well :D |
| 06:28.58 | benjk | and our BRI is EuroISDN |
| 06:29.00 | JT | yeah, japan is weird like that |
| 06:29.20 | JT | do you run a-law or Mu-law over BRI there? |
| 06:29.28 | benjk | so if our BRI wasn't using ulaw instead of alaw, then the codecs between PRI and BRI wouldn't match |
| 06:29.37 | benjk | that's not something you want to do on a national level |
| 06:29.38 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 06:30.25 | benjk | NTT also has an E1 derived PRI spec which was never implemented, but it also uses ulaw instead of alaw |
| 06:30.40 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 06:31.00 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 06:31.02 | tengulre | hi,all |
| 06:31.09 | JT | hmm |
| 06:31.14 | JT | is that the J1? |
| 06:31.26 | benjk | Japanese PRI is T1 but different framing == J1 |
| 06:31.36 | JT | ah right |
| 06:31.43 | benjk | and Japanese BRI is EuroISDN with ulaw instead of alaw |
| 06:31.54 | tengulre | how to setting h323 in asterisk? I want using windows's netmeeting to dial my sip phone ? but I dont know how to setting the 'gatekeeper'? |
| 06:31.57 | JT | same number of channels in CAS and CCS modes respectivly? |
| 06:32.09 | benjk | <PROTECTED> |
| 06:32.24 | benjk | yes, all the same, just the framing is different |
| 06:32.54 | tengulre | but netmeeting how to register to asterisk with gatekeeper? |
| 06:32.55 | JT | frame synch or something like that? |
| 06:33.04 | benjk | so if you have a T1/J1 controller chip that does the framing transparently, then from your end it just looks like a T1 |
| 06:33.24 | benjk | and many controllers do this these days |
| 06:34.54 | benjk | however, NTT has implemented a whole bunch of things in the ISDN specs that other telcos didn't |
| 06:35.07 | benjk | and that's where it can become incompatible |
| 06:35.11 | benjk | especially with Asterisk |
| 06:35.43 | benjk | because many things ISDN does and can do, libpri (and thereby Zaptel/Asterisk) simply ignores |
| 06:36.11 | *** join/#asterisk tengulre11 (n=tengulre@61.185.224.66) |
| 06:36.19 | benjk | I remember a comment by kapejod somewhere in the source code that hit the nail on the head |
| 06:36.29 | tengulre11 | anybody using h323? |
| 06:36.47 | benjk | it said something like "ISDN is more than a bunch of analog channels with digital codecs" |
| 06:37.19 | JT | true |
| 06:37.54 | benjk | tenguire, I don't think you need h323 |
| 06:38.07 | benjk | NetMeeting does support SIP these days, or so I heard |
| 06:38.15 | JT | so i bought some flat 8core modular cable and 8P8C plugs to suit, to make my T1 cable |
| 06:38.18 | JT | then i got home |
| 06:38.19 | JT | and thought |
| 06:38.23 | JT | why did i buy that? |
| 06:38.36 | Grnd-Wire | hmm.. A network cable woulda worked well :D |
| 06:38.42 | benjk | heh |
| 06:39.00 | JT | Grnd-Wire: you mean cat5... ethernet crossover cable does not do T1 crossover |
| 06:39.05 | benjk | but it helps if there is some trouble and your customer asks "who made that cable?" |
| 06:39.42 | JT | but yeah, clearly T1 uses differential signals, so there is no point to using flat cable |
| 06:39.54 | benjk | I always carry a set of NTT supplied T1 cables with me for that purpose |
| 06:39.58 | Grnd-Wire | JT: Ya, know.. 1,2,4,5 .. I didn't know he needed crossover.. |
| 06:40.14 | Grnd-Wire | It's hard to make crossover cables out of flat cable, cause you can't really cross them :P |
| 06:40.21 | JT | i must've for some reason thought i heard someone say to use flat cable |
| 06:40.26 | JT | i'm sure it's possible |
| 06:40.29 | benjk | same why I use IBM boxes for demos |
| 06:40.41 | benjk | the label on the cable or box is most important over here |
| 06:40.59 | JT | benjk: ntt cables, are they Cat5 UTP? |
| 06:41.28 | Grnd-Wire | JT: Oh yes, you can use "silver-satin" for T1.. but only if you're going to connect a NIU to the CSU, since it won't be crossed.. |
| 06:41.29 | benjk | presumably they are, but NTT says they are INS1500 cables |
| 06:41.40 | benjk | so that's what they are in the minds of customers |
| 06:42.01 | JT | Grnd-Wire: silver satin? |
| 06:42.16 | benjk | Moody Blues cabling |
| 06:42.27 | benjk | no wait, that was white satin |
| 06:42.31 | Grnd-Wire | HAHA.. You know, everything needs a cute name.. |
| 06:42.51 | JT | what's the distance limit on using flat cable? |
| 06:42.56 | benjk | yeah, like I saw this truck today |
| 06:42.59 | Grnd-Wire | That flat grey cable.. You just call it silver satin and everyone (except you obviously) knows that I'm talking about :D |
| 06:43.09 | benjk | it said Crystal Clara |
| 06:43.15 | benjk | never heard of that brand before |
| 06:43.27 | JT | it's cremey-beige usually, here |
| 06:43.32 | benjk | but it kinda sounds familiar :) |
| 06:44.01 | Grnd-Wire | JT: hmm - Tricky question.. I know it's REALLY far.. You want twists though.. If you're going more than a couple a hundred feet, you've gotta have them configure the NIU for that anyway.. |
| 06:44.25 | Grnd-Wire | I know I've never extended a smart jack with anything other than CAT5, cause it'd be crazy to do anything else . :P |
| 06:44.28 | JT | Grnd-Wire: yeah i gathered it would be "far" with twists, i was curious without |
| 06:45.06 | Grnd-Wire | JT: Considering what kind of cable is buried in the ground.. T1's are differential, so they're extremely tolerant of noise and such.. |
| 06:45.19 | JT | yes |
| 06:45.34 | benjk | all new T1s (ahem J1s) in Japan come over fiber |
| 06:46.03 | Grnd-Wire | So maybe I can interject a question in the middle of all of this.. Has anyone ever setup any sort of asterisk peering? |
| 06:46.09 | Grnd-Wire | err.. I guess it's called "friends" |
| 06:46.20 | Grnd-Wire | I'm looking to play with DUNDI, and I'm not sure where to start.. |
| 06:46.23 | benjk | any sort of asterisk peering sounds very broad though |
| 06:46.43 | JT | benjk: are the T1 cables you see STP or UTP? |
| 06:46.52 | benjk | you'll find that anybody habving done asterisk setups will have done some kind of peering |
| 06:47.06 | benjk | UTP |
| 06:47.08 | JT | all the ISDN BRI S-bus cables I see around here are STP |
| 06:47.13 | Grnd-Wire | benjk: At the moment I have two working machines that don't know about each other.. when I'm done I want them to know about each other, and know that one machine hosts 2xxx extensions, and the other hosts 3xxx extensions.. |
| 06:47.28 | Grnd-Wire | benjk: Is that a little more specific for ya? :P |
| 06:47.54 | benjk | you can use the switch statement in the dialplan for that |
| 06:48.16 | JT | benjk: any observations, re: s-bus cables? |
| 06:48.38 | benjk | I am not into cable science really |
| 06:48.50 | JT | ah ok |
| 06:49.03 | JT | well it's obvious to me because of the shielded modular plugs |
| 06:49.09 | benjk | I use what cables come with the equipment or what generally accepted suppliers ship |
| 06:49.13 | JT | all the metal sheeting around them |
| 06:49.39 | Grnd-Wire | benjk: hmm.. Switch? That's not in the O'Reilly book.. at least in the Application reference. Is that where I should be looking? |
| 06:49.55 | benjk | I never read that book |
| 06:50.29 | benjk | you may want to search for dialplan and switch cmd at Voip-info.org |
| 06:50.50 | benjk | basically it is a sort of import of another asterisk server's dialplan (or part thereof) |
| 06:51.36 | benjk | and for transport, you could use IAX trunking |
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| 06:51.59 | *** mode/#asterisk [+o Corydon76-home] by ChanServ |
| 06:52.04 | benjk | set up a peer and friend in each machine's iax.conf |
| 06:52.42 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 06:52.57 | benjk | peer entry of machine A corresponds to friend entry of machine B and vice versa |
| 06:54.09 | benjk | if you have multiple concurrent calls between the two machines you can enable IAX trunking in those entries and they will then bundle all the calls between them into a single data stream |
| 06:54.22 | Grnd-Wire | ok, that's what I was looking for.. So there isn't a specific way to setup trunking.. OOOH, ok.. |
| 06:54.42 | Grnd-Wire | Now THAT is what I'm talking about. :) Very cool.. ok - Let me go read some more. ;) |
| 06:54.43 | benjk | well, like I said, you need to have those corrsponding entries in iax.conf |
| 06:55.03 | benjk | and for trunking those entries need to have a line trunk=yes |
| 06:55.09 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 06:55.12 | Grnd-Wire | gotcha |
| 06:55.33 | tengulre | HI,all |
| 06:55.37 | Grnd-Wire | So what's a good IAX softphone to use for testing? |
| 06:55.43 | tengulre | anybody can help me? |
| 06:55.49 | benjk | actually it is peer and user |
| 06:55.51 | benjk | not friend |
| 06:56.05 | benjk | friend is a way to combine both entries into a single entry |
| 06:56.08 | tengulre | Grnd-Wire, iaxcomm |
| 06:56.23 | tengulre | anybody using H323 in asterisk? |
| 06:57.15 | benjk | some folks like to use friend, but I only use that for terminals, not for peering between servers |
| 06:57.41 | benjk | some people get religious about the friends versus user/peer thing |
| 06:57.43 | Grnd-Wire | benjk: Ok, so you do NOT use friend? You're recommending against it.. ok That's cool - Until I know otherwise, I'll work on that understanding myself |
| 06:57.50 | JT | benjk: reasoning? |
| 06:57.58 | benjk | I dunno |
| 06:58.05 | Grnd-Wire | benjk: I'm sure they do.. I'll just do it your way, cause you're talking to me.. :) |
| 06:58.27 | tengulre | JT, do u using H323 in asterisk? how to using netmeeting with h323 gatekeepter in asterisk? |
| 06:58.34 | JT | tengulre: nup |
| 06:58.36 | benjk | I personally find it easier to keep things clear by separating the incoming from the outgoing connections in separate entries |
| 06:58.48 | JT | right |
| 06:59.00 | tengulre | how to using netmeeting with h323 gatekeepter in asterisk? |
| 06:59.06 | tengulre | SOS! |
| 06:59.19 | benjk | if you use friends, then there are some things that are implicit, not explicit |
| 06:59.32 | benjk | so its less to type, but easy to overlook something |
| 06:59.39 | JT | tengulre: ffs, you've asked enough already, perhaps read up on it now since no-one here knows |
| 06:59.59 | benjk | he doesn't even need H323 for NetMeeting |
| 07:00.14 | tengulre | benjk: do u know how to do ? |
| 07:00.23 | e-ddie | are there any good video softphones for linux, with h.263 support? |
| 07:00.42 | benjk | all I know is that the Linux clone for NetMeeting (forgot the name) has changed from H323 to SIP |
| 07:00.51 | benjk | if they did that, it means MS changed to SIP |
| 07:01.08 | e-ddie | ekiga is the name of it |
| 07:01.16 | benjk | did they change the name? |
| 07:01.25 | benjk | it was something with meeting at the end |
| 07:01.33 | tengulre | benjk: my mean is how to using NetMeeting to register to asterisk with gatekeeper? |
| 07:01.51 | e-ddie | was gnomemeeting before |
| 07:01.56 | Grnd-Wire | benjk: http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers Lookey what I found :D |
| 07:02.00 | benjk | ah yes , that's the name |
| 07:02.15 | benjk | ;) |
| 07:02.53 | tengulre | nobody know? |
| 07:02.56 | JT | tengulre: and the point is you don't have to if netmeeting can use SIP |
| 07:03.10 | JT | tengulre: have you checked and are 100% absolutely usre that it cannot do SIP? |
| 07:03.15 | JT | s/usre/sure/ |
| 07:03.38 | tengulre | JT: :( |
| 07:04.17 | tengulre | my mean is Netmeeting ---> <H323>---->Asterisk ---><SIP>--->sip phone |
| 07:04.34 | JT | i understand that, but do you need to use H.323? |
| 07:04.42 | JT | SIP would be a lot easier |
| 07:04.51 | JT | H.323 requires a gatekeeper |
| 07:05.15 | tengulre | or NetMeeting ----> <H.323>--->Asterisk gatekeeper |
| 07:05.28 | Juggie | ~seen theplot |
| 07:05.33 | jbot | theplot <i=ThePlot@202.164.38.210> was last seen on IRC in channel #asterisk, 14d 13h 6m 42s ago, saying: 'I did set in the address field to match the username too'. |
| 07:05.41 | JT | or NetMeeting ---> <SIP> ---> Asterisk |
| 07:05.47 | JT | why don't you want to consider it? |
| 07:05.48 | tengulre | JT: no |
| 07:06.06 | JT | why not? |
| 07:06.29 | tengulre | JT, because the NetMeeting can not support SIP. |
| 07:07.55 | tengulre | anyway Netmeeing how to register to asterisk with gatekeeper???? |
| 07:08.25 | JT | adding extra question marks won't add extra answers |
| 07:08.26 | JT | anyway |
| 07:08.32 | JT | seems you are living in the past |
| 07:08.34 | JT | With Windows XP, Microsoft dropped continued development of H.323-based Netmeeting in favor of the SIP-based collection of standards, discussed below. Microsoft has explicitly embedded SIP within its new .Net framework, and SIP is used for its Windows Messenger product (versus H.323 for its MSN Messenger product). |
| 07:09.30 | tengulre | JT, Thanks! |
| 07:09.47 | JT | http://66.102.7.104/search?q=cache:JijJj6bFLToJ:www.eng.mu.edu/rehab/Rehab167/Mod3/teleconf/h323-sip.htm+netmeeting+sip&hl=en&gl=au&ct=clnk&cd=1&ie=UTF-8 |
| 07:13.15 | tengulre | JT, I can not open it! |
| 07:15.55 | *** join/#asterisk vlt (n=dm@p54B344D4.dip0.t-ipconnect.de) |
| 07:16.16 | Grnd-Wire | benjk: hmm - I don't think switch is what I want, since it says you can't use switch in a circular fashion.. |
| 07:16.24 | Grnd-Wire | "Notes: You may not establish circular links by switching serverA to serverB and serverB to serverA!" |
| 07:17.04 | JT | tengulre: try http://www.eng.mu.edu/rehab/Rehab167/Mod3/teleconf/h323-sip.htm |
| 07:18.20 | tengulre | JT,Thank you very much! |
| 07:20.05 | JT | google for netmeeting sip |
| 07:20.13 | JT | heaps of things talk about ms dropping h.323 |
| 07:20.20 | JT | and going to ms messenger |
| 07:22.04 | Un1x | anyone do AGI scripting for money :D? |
| 07:32.10 | Grnd-Wire | oh god - This is so cool.. It's actually working properly.. Well.. I should say, I'm seeing diagnostic messages showing up :D |
| 07:36.40 | benjk | Un1x, didn't you talk to asteriasgi? |
| 07:37.01 | Un1x | no whos that? |
| 07:37.07 | Grnd-Wire | benjk: What does this error mean? Aug 21 00:36:42 NOTICE[4973]: chan_iax2.c:7303 socket_read: Rejected connect attempt from 192.168.4.200, request '000@iax2users' does not exist |
| 07:37.22 | benjk | SwK said they already have an AGI or app that did exactly what you wanted and he said you should contact them on Monday |
| 07:37.35 | Grnd-Wire | benjk: It's obvious the dial plan stuff is working, cause that is happening when I dial from the other console.. |
| 07:37.36 | benjk | I think it was sales@asteriasgi.com |
| 07:38.29 | benjk | if they don't have what you need, contact me again |
| 07:38.42 | benjk | Sunrise-tel.com |
| 07:38.46 | benjk | enquiries |
| 07:38.54 | Un1x | ok |
| 07:39.10 | Un1x | i gotta go sleep now tho |
| 07:39.12 | Un1x | kinda late |
| 07:39.19 | Un1x | thanks alot tho benjk have a goodnite |
| 07:39.20 | benjk | but asteria are good at what they do, so I am confident that you'll find their stuff useful |
| 07:39.21 | Un1x | cya |
| 07:39.32 | Un1x | ok goodnight and thanks alot man :) |
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| 07:40.43 | benjk | Grnd-Wire, I have to go to see a customer before end of business and its 4:40 pm here, contact me later if you still have problesm |
| 07:41.04 | Grnd-Wire | benjk: Sure.. Have a good, err.. night! :D |
| 07:41.10 | Grnd-Wire | I appreciate the help you've given me.. |
| 07:41.12 | vlt | Hello. Very strange (for me) NAT behavior here: Asterisk is behind NAT, "extenip" is set correctly in sip.conf. After my external IP changes (every night) Asterisk is still working for a few hours *without* correcting "externip=". When I finally change sip.conf and reload/restart it takes up to 1 hour until it works again. "sip show registry" shows "Request Sent" (for two different SIP registrations). How can I speed up the register process? |
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| 08:05.04 | Grnd-Wire | boobee2: hmm - Some sort of conditional using the AsterDB ? |
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| 08:10.09 | vlt | Sorry for posting again (my connection was reset) but ... |
| 08:10.10 | vlt | Hello. Very strange (for me) NAT behavior here: Asterisk is behind NAT, "extenip" is set correctly in sip.conf. After my external IP changes (every night) Asterisk is still working for a few hours *without* correcting "externip=". When I finally change sip.conf and reload/restart it takes up to 1 hour until it works again. "sip show registry" shows "Request Sent" (for two different SIP registrations). How can I speed up the register process? |
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| 08:11.02 | boobee2 | Grnd-Wire with some kind of - CLI> database put var value - ? |
| 08:11.15 | Assid | vlt: get a dynamic host name.. and have that update .. then keep using sip reload |
| 08:11.25 | ChrisDE4 | hi... whats this: channel.c: Got a FRAME_CONTROL (3) frame on channel ... |
| 08:11.25 | Juggie | vlt, use externhost rather then externip |
| 08:11.32 | Juggie | and setup a dynamic dns somewhere. |
| 08:11.40 | Assid | didnt i just say that |
| 08:11.48 | ChrisDE4 | ...WARNING[5854] channel.c: Unable to handle indication 3 for 'SIP/ |
| 08:12.22 | boobee2 | Grnd-Wire what would you suggest using to write/read that var in astdb from a php page? |
| 08:12.52 | Assid | boobee2: why not use SQL ? |
| 08:13.40 | boobee2 | Assid is that stable? setup will be ~500 sip peers loaded |
| 08:14.25 | boobee2 | sorry, i read SQL = mysql |
| 08:14.38 | Assid | welll you could use mysql too |
| 08:15.07 | Assid | thats how realtime works |
| 08:15.09 | boobee2 | you mean storing sip.conf in a sql db? realtime isn't it? |
| 08:15.12 | boobee2 | ok |
| 08:15.35 | boobee2 | i was working that way, but had been noticed of unstability of realtime ATM so i stopped |
| 08:15.58 | Assid | unstability? |
| 08:16.56 | boobee2 | yup, somebody told me bout it here on friday |
| 08:17.21 | boobee2 | but was just general talking |
| 08:18.08 | boobee2 | anyway, i'll test it myself so i'll be fixed |
| 08:19.05 | *** join/#asterisk Vec (n=Vector@dsl-146-66-58.telkomadsl.co.za) |
| 08:19.15 | *** join/#asterisk ghenry (n=ghenry@suretecsystems.plus.com) |
| 08:19.52 | *** join/#asterisk webman (n=adamg@52.87.233.220.exetel.com.au) |
| 08:20.07 | Vec | Is it possible to run 2 high density analogue digium cards without problems? I have heard that slow linux interupts cause problems ? |
| 08:20.39 | *** join/#asterisk pa (n=Paolo@unaffiliated/pa) |
| 08:20.40 | Assid | well. try and avoid transcoding.. that would help you plenty |
| 08:21.12 | Vec | Assid : Do you mean changing from one codec to another between cards ? |
| 08:21.19 | JT | try to make sure each card has an interupt that is unshared |
| 08:21.31 | vlt | Assid: Thank you. What would a dyndns name be useful for? Should I set "externhost=" to that dyndns-addr? Where is the difference to updating the IP manually in sip.conf (I mean, why doesn't it work immediately)? |
| 08:21.37 | Assid | yeah.... between devices.. |
| 08:21.50 | Vec | Has anyone got 2 cards to work well together ? |
| 08:21.54 | Assid | vlt: not sure.. i use externost |
| 08:21.58 | ChrisDE4 | again.. does anyone know what "Got a FRAME_CONTROL (3) frame on channel.." means? ... is this a synonym for "Hangup"? |
| 08:22.37 | Grnd-Wire | Assid: I'm trying to get IAX trunking to work between two boxes. I'm following the directions on voip-info.org, but I'm getting an error message - and it's not clear to me what it means.. Even after turning on IAX debugging. |
| 08:22.51 | Grnd-Wire | Anyone interested in seeing the error? :D |
| 08:22.59 | Assid | what error do you get in the CLI on both the boxes |
| 08:23.13 | Assid | in the one sending the call... and the one receiving the call |
| 08:23.33 | Grnd-Wire | Assid: It says it's rejecting the call because: "000@iax2users does not exist" |
| 08:23.51 | Grnd-Wire | the iax2users is the context I'm placing the call into on the PEER (receiving) |
| 08:24.47 | *** join/#asterisk sskyles (n=Steve@237.150.119.70.cfl.res.rr.com) |
| 08:25.26 | vlt | Assid: Now I set "externhost=a.domain.name". Turning on "sip debug" tells me that outgoing REGISTER requests use my external IP (not the hostname) but it still doesn't work. |
| 08:25.26 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 08:26.27 | *** join/#asterisk BugKham (n=bugkham@ppp-58.8.2.230.revip2.asianet.co.th) |
| 08:27.15 | BugKham | anyway to identify a call from my E1 if there's no callerid? |
| 08:31.33 | vlt | I just added another "register =>" line to sip.conf (a peer I haven't used for at least 3 days), reloaded, but it only shows "Request Sent" ... Could it be a problem on the NAT router (Debian Sarge)? |
| 08:33.16 | JT | BugKham: magic? |
| 08:33.28 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 08:33.28 | Grnd-Wire | Assid: You still interested in those error messages? I've got them now.. |
| 08:33.40 | *** part/#asterisk ChrisDE4 (n=ChrisDE@88.128.40.29) |
| 08:34.29 | webman | can anyone see anything wrong in this fragment from a sip invite "m=audio 0 RTP/AVP 18 8 100" |
| 08:34.30 | BugKham | JT: just wondering if there's a way |
| 08:35.03 | Grnd-Wire | ok, well I'll post these for anyone else :) The USER (person dialing) is showing me this error: Aug 21 01:32:56 WARNING[4973]: chan_iax2.c:7075 socket_read: Call rejected by 192.168.4.200: No such context/extension |
| 08:35.06 | webman | BugKham: you could ID it by the uniqueid, or the channel perhaps |
| 08:35.35 | Grnd-Wire | And the PEER is showing: *CLI> Aug 21 00:35:02 NOTICE[6731]: chan_iax2.c:7303 socket_read: Rejected connect attempt from 192.168.4.53, request '000@iax2users' does not exist |
| 08:36.03 | Grnd-Wire | I don't know what it's complaining about! The context exists.. what is that 000@iax2users all about? |
| 08:36.16 | BugKham | webman: but that will not tell where the call comes from anyway |
| 08:36.53 | *** join/#asterisk pnlarsson (n=niklas@c83-248-0-248.bredband.comhem.se) |
| 08:37.04 | BugKham | webman: any other identity from telco, apart from callerid? |
| 08:38.10 | webman | BugKham: well, you didn't say you wanted to know where it was *from*.... you could ask the telco to supply CID, and reject any calsl without CID |
| 08:38.25 | *** join/#asterisk Skaag (n=hintza@212.199.180.157.static.012.net.il) |
| 08:38.42 | webman | Grnd-Wire: what is the dial line on the user box |
| 08:39.02 | Skaag | I'm trying to connect xten lite to asterisk, it says login timed out. |
| 08:39.17 | JT | BugKham: identify their voice? their pretty much is no other way |
| 08:39.21 | JT | unless you're a telco |
| 08:39.23 | *** join/#asterisk Arno[Slack] (i=100@master.infinityperl.org) |
| 08:39.28 | JT | and have SS7 interconnect with them |
| 08:39.32 | JT | then you could get ANI |
| 08:40.44 | Skaag | oh god, ss7..! |
| 08:40.55 | Skaag | I remember 10 years ago I played with SS7 ;-) |
| 08:41.00 | JT | heh |
| 08:41.03 | Skaag | and with the earlier ITUT protocol |
| 08:41.28 | Skaag | boxing the world with tonelock ;-) |
| 08:41.33 | Skaag | and TLO |
| 08:41.41 | Skaag | (The Little Operator) |
| 08:42.16 | Skaag | And today, I can't connect a simple SIP client with Asterisk ;-) |
| 08:43.39 | Grnd-Wire | webman: It's simple, just: exten => _7XXX,1,Dial(IAX2/to-andy/${EXTEN:1},30,r) |
| 08:43.59 | BugKham | JT: ok |
| 08:44.09 | Grnd-Wire | webman: And I've got all the credentials listed in my iax.conf file - I can send you to pastebin if you want, none if it is confidential? |
| 08:44.46 | webman | Grnd-Wire: well, that is where it is getting the 000 from I suppose (I assume you are dialling 7000) |
| 08:45.02 | webman | what if you add the context in the dial line? |
| 08:45.09 | Grnd-Wire | oh god - I just figured that out myself :D |
| 08:45.15 | Grnd-Wire | THAT is what that :1 means.. drop the 1.. |
| 08:45.18 | Grnd-Wire | err.. One digit |
| 08:45.19 | Grnd-Wire | FUCK ME |
| 08:45.41 | Skaag | It seems like my Asterisk is trying to contact my sip client back, but there's no way it's going to do that through the firewall |
| 08:45.45 | Grnd-Wire | You know - that's what I get for using example code, and not completely understanding it. :D |
| 08:45.46 | Skaag | how does that work then? |
| 08:46.01 | Grnd-Wire | webman: yay.. It does what it's supposed to now. :D |
| 08:46.24 | webman | skaag: the client sends out the data, so the firewall should know to let the data back in again... |
| 08:46.34 | webman | Grnd-Wire: glad to help :) |
| 08:47.13 | Skaag | webman: but it looks like the firewall initiates new connections, it doesn't reply on the same initial client connection |
| 08:47.19 | Skaag | I don't see why the firewall would agree to allow that |
| 08:47.38 | Skaag | unless i'm wrong in my interpretation of how this works |
| 08:48.20 | webman | skaag: yep, you just might be wrong :) since it is working for millions of people out there .... |
| 08:48.53 | webman | skaag: sometimes the firewall will understand SIP and be able to do extra smart things, but it should work fine without this anyway |
| 08:49.41 | Skaag | I must be doing something wrong then |
| 08:50.44 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 08:52.27 | Grnd-Wire | ok guys - Time for bed.. Thanks for the help! heh.. |
| 08:55.40 | *** join/#asterisk E-bola (i=bola@217.147.82.8) |
| 08:55.59 | E-bola | Do anybody run asterisk on a multihomed server? |
| 08:56.49 | Vec | Anyone had any experiance running 2 or more digium cards in a linux box, I have heard that it does not work very well ? |
| 08:58.08 | E-bola | Multihomed as in running an asterisk on a server with 2 nic's a wan nic and a lan nic |
| 09:02.35 | *** join/#asterisk pyrom (n=pyro@86.84-48-44.nextgentel.com) |
| 09:02.55 | pyrom | Having some register issues, with my sipura 3102 and asterisk, what's best method of debugging? |
| 09:03.12 | pyrom | using sip debug and asterisk -cvvvv |
| 09:03.20 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:03.57 | pyrom | ..."register_verify: Peer 'pstn-spa3k' is trying to register, but not configured as host=dynamic |
| 09:03.57 | pyrom | "..."Registration from 'pstn-spa3k <sip:pstn-spa3k@10.0.0.3>' failed for '10.0.0.40'" Where 10.0.0.40 is the sipura |
| 09:04.56 | *** part/#asterisk sskyles (n=Steve@237.150.119.70.cfl.res.rr.com) |
| 09:08.31 | *** join/#asterisk ltd (n=z@202-161-16-50.dyn.iinet.net.au) |
| 09:13.24 | *** join/#asterisk ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 09:14.08 | *** join/#asterisk atapi2 (n=virgill4@c-69-180-119-156.hsd1.fl.comcast.net) |
| 09:15.53 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 09:16.04 | *** join/#asterisk kore (i=kore@mindwipe.org) |
| 09:16.29 | *** part/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 09:18.11 | *** join/#asterisk kannan (n=kannan@125.22.67.231) |
| 09:18.28 | kannan | hello all |
| 09:18.58 | *** join/#asterisk SanketMedhi (n=sanketme@221.135.151.62) |
| 09:19.19 | SanketMedhi | !topic |
| 09:20.22 | *** join/#asterisk Ahrimanes (n=michael@81.7.159.2) |
| 09:20.29 | E-bola | hi |
| 09:20.34 | Ahrimanes | morning |
| 09:22.51 | kannan | thanx Jeffjohnson, any link I can learn from on howto? |
| 09:23.21 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 09:23.43 | JT | you really need to provide a little more information on what you want to do, kannan |
| 09:24.05 | SanketMedhi | hello, I want to know if Realtime works perfectly with Asterisk |
| 09:24.22 | Ahrimanes | SanketMedhi: define perfectly ? |
| 09:24.24 | SanketMedhi | I have faced probs with sip.conf and extensions.conf thru Realtime |
| 09:24.37 | SanketMedhi | Ahrimanes: was that fine? |
| 09:24.38 | Jeffjohnson | kannan: no :) you must spawn the asterisk gui on an serial device, /etc/inittab would help. Should be the same way like connecting over serial to a console |
| 09:24.50 | Ahrimanes | SanketMedhi: more accurate, good yes |
| 09:25.15 | SanketMedhi | Ahrimanes: good yes? |
| 09:25.21 | kannan | oh ok, I am currently using the gnudialer and vicidial to make out bound calls thru SIP termination, I'd like to interface now thru PSTN and call a telco's number thru the hyperterminal to input old telephone number, the telco will send me back the new telephone number |
| 09:25.25 | SanketMedhi | Ahrimanes: is that the answer? |
| 09:25.29 | Ahrimanes | SanketMedhi: yep... but it's working nicely here though |
| 09:25.52 | *** join/#asterisk Sonderblade (n=mah@static-213.131.147.169.addr.tdcsong.se) |
| 09:26.12 | Jeffjohnson | kannan: ok you don't want the asterisk interface over serial .p |
| 09:26.17 | Ahrimanes | anyone know an enterprise grade pc switchboard solution for asterisk, that can integrate with exchange to see if people are in meetings etc? |
| 09:26.35 | SanketMedhi | Ahrimanes: so you mean I can dump some data into a Mysql DB and Asterisk Realtime will pick it up without any need to reload/restart on the fly? |
| 09:28.16 | Ahrimanes | SanketMedhi: yes, we're doing that with sip peers |
| 09:28.18 | *** join/#asterisk docelmo (n=vircuser@55-65.126-70.tampabay.res.rr.com) |
| 09:29.57 | SanketMedhi | Ahrimanes: Ok, a friend also says SIP peers works fine, but what about extensions? |
| 09:31.33 | *** join/#asterisk grEvenX (n=even@pc100-15.ktv.no) |
| 09:33.43 | kannan | ok, i get that spawn on a serial device , i'll try that out . Whats the equivqlent of hyper terminal on a linux box? :) |
| 09:33.48 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 09:35.45 | SanketMedhi | Ahrimanes: ? |
| 09:37.08 | Jeffjohnson | kannan: minicom |
| 09:37.32 | Vec | Anyone had any experiance running 2 or more high density digium cards in a linux box, I have heard that it does not work very well ? |
| 09:38.02 | Jeffjohnson | kannan: but i thought you want to dial numbers? I dont really know what you want :) |
| 09:38.09 | Ahrimanes | Vec: there tends to be some problems with interrupts |
| 09:38.11 | *** join/#asterisk key2 (n=key2@251.9.39-62.rev.gaoland.net) |
| 09:39.07 | SanketMedhi | are there any compatibility issues between softphones? eg. ekiga and x-lite? |
| 09:39.43 | kannan | thanx again Jeffjohnson :) ; I do need to dial out to one telephone number provided by the telco. If the hyperterminal calls this no, we can input any number and it will respond with the changed number |
| 09:40.13 | E-bola | kannan: what kinda weird ass setup is that?= |
| 09:40.28 | *** join/#asterisk vlrk (n=root@202.65.134.119) |
| 09:40.39 | kannan | so I was thinking I can interface thru 8 WLL phones which come with modem |
| 09:41.14 | Jeffjohnson | kannan: my asterisk skills are too low :) don't know if the asterisk cli is what you really want |
| 09:41.37 | kannan | heh, I have a huge database of old numbers and a few yellow pages are ready to buy this if I can get the changed numbers also after getting the new number and then doing a telephonic survey |
| 09:45.02 | kannan | a large number of telephone exchanges just upgraded from manual to electronic in many towns. The telco company has provided a number which we can call and input the old number, it will then respond with a new number, I was wondering if it is possible to somehow do this on an autodialler like gnuialer or the astguiclient application. |
| 09:45.46 | kannan | the number can be called only by hyperterminal (or minicom? maybe :)) |
| 09:46.53 | pyrom | minicom would ofc work |
| 09:47.08 | kannan | ty pyrom. |
| 09:47.31 | tzafrir | kannan, minicom/hyperterminal is for a serial console. you can connect to a linux server via a serial terminal using getty and a serail port |
| 09:47.49 | tzafrir | However if you connect via a network, use ssh and putty |
| 09:47.58 | pyrom | but do you use some modem? |
| 09:48.14 | kannan | the dialler came with scratch install instructions , so i felt a bit like like i knew asterisk, but actually i dont i guess , ha hah |
| 09:48.16 | vlrk | i could not able to trace where we need to give the user /password ,mysql host for the mysql authentication purpose , i want to use the voicemail with the mysql database . I think extconfig.conf is the place where we mention the datbase driver name , database name and the table name . so can any body give an hint where can i find my required details |
| 09:48.33 | kannan | i can manually do it thru a wll phone thru which i am connected to the net |
| 09:49.05 | zaswk | hi, i'am using a tdm22 card, FXS line is working with a phone but not with a fax/phone device (no ring, but i can call, and even answer): the second one is not ringing, so it is unable to answer and receive fax. Any idea ? |
| 09:49.26 | Vec | Ahrimanes : I have also heard there is problems with interupts, has anyone sorted those problems out ? |
| 09:49.50 | kannan | currently i use only SIP ATAs (fxs ports) and a far end termination provider terminates to pstn for me |
| 09:50.42 | kannan | the WLL phone has in-built modem |
| 09:51.12 | zaswk | Vec: digium cards have to be alone on one interrupt line, else i experienced bad things, they really dislike any sharing (at least here) |
| 09:52.33 | *** join/#asterisk somegeek_ (i=levin@tor/regular/somegeek) |
| 09:54.44 | zaswk | Vec: i had to disable Pnp and force irqs in BIOS (beware of PCI slots too, e.g. ASUS motherboards 1/5 2/6) |
| 09:55.13 | Ahrimanes | Vec: basically if you KNOW that you put the cards on seperate pci busses etc, it should be possible to run more cards in one server.. but not sure it's a good idea |
| 09:56.05 | *** join/#asterisk tengulre (n=tengulre@222.90.66.4) |
| 09:56.33 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 09:57.31 | zaswk | Ahrimanes: any idea about why a phone is ringing while another is not on an FXS line ? |
| 09:58.13 | Ahrimanes | zaswk: huh? |
| 09:59.17 | Kasimeng | Does anyone know about Asterfax? |
| 09:59.29 | Un1x | join #asterfax |
| 09:59.30 | Un1x | lol |
| 09:59.36 | Kasimeng | thx.. |
| 09:59.43 | Un1x | im joking dude! |
| 09:59.56 | Kasimeng | ..... |
| 10:00.08 | Kasimeng | then, does anyone use it before? |
| 10:00.53 | Kasimeng | I mean use it with trixbox |
| 10:00.54 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 10:01.06 | zaswk | Ahrimanes: i connected a phone on a FXS line (tdm22b port 1) -> ok, then i connected a Sharp ux370 phone/fax -> no ring, but everything else is working. I suspected signalling, tried fxo_ls,fxo_ks but without success |
| 10:04.25 | Ahrimanes | zaswk: ah.. hm no not really, i mostly have experience with ata's for connecting phones and faxes |
| 10:05.10 | zaswk | Ahrimanes: ok :) |
| 10:06.01 | *** join/#asterisk FlatFoot (n=simon@80.88.192.113) |
| 10:09.22 | *** part/#asterisk vlrk (n=root@202.65.134.119) |
| 10:21.05 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 10:21.29 | xnon | friends exist any repositorie for install freepbx in debian with apt? |
| 10:22.13 | xnon | i have debian sarge and asterisk but i cant install freepbx |
| 10:22.36 | Ahrimanes | xnon: http://www.google.dk/search?hs=GBR&hl=da&client=firefox&rls=org.mozilla%3Aen-US%3Aunofficial&q=freepbx+deb&btnG=S%C3%B8g&meta= |
| 10:22.47 | xnon | Connecting to database..FAILED |
| 10:22.47 | xnon | [FATAL] mysql PHP libraries not installed |
| 10:26.06 | xnon | i have a freepbx package but i cant install but this error is show me! |
| 10:26.09 | xnon | :( |
| 10:26.40 | xnon | the first time i wasnt have this error but latter show it |
| 10:26.51 | SanketMedhi | xnon: you don't have the php-mysql package installed |
| 10:27.15 | xnon | really? |
| 10:27.36 | xnon | what is the exactly name of this package? |
| 10:28.26 | xnon | i have a mysql and php installed friend |
| 10:28.58 | xnon | ii php4-mysql 4.3.10-16linex MySQL module for php4 |
| 10:29.09 | xnon | ii php4 4.3.10-16linex server-side, HTML-embedded scripting languag |
| 10:29.26 | xnon | ii mysql-server-4 4.1.11a-4sarge mysql database server binaries |
| 10:29.43 | xnon | you see it |
| 10:29.44 | xnon | ? |
| 10:30.02 | xnon | or the package is the other one'? |
| 10:33.07 | *** join/#asterisk ltd (n=z@202-161-16-50.dyn.iinet.net.au) |
| 10:34.53 | RoyK | xnon: /j #freepbx |
| 10:34.54 | SanketMedhi | xnon: where is apache? |
| 10:35.12 | SanketMedhi | umm yeah :P |
| 10:35.51 | SanketMedhi | can someone tell me about compatibility issues between soft phones? |
| 10:36.07 | SanketMedhi | ekiga and x-lite don't seem to work with each other |
| 10:36.12 | SanketMedhi | any clues? |
| 10:38.31 | Dico_ | SanketMedhi, what do you mean they don't seem to work together / |
| 10:38.40 | Dico_ | you mean on the same computer ? |
| 10:38.42 | *** join/#asterisk saftsack (n=saftsack@p54A7D3C2.dip.t-dialin.net) |
| 10:40.09 | SanketMedhi | Dico_: no on two different machines |
| 10:40.36 | SanketMedhi | Dico_: I have an Asterisk server and 2 clients, one Windoze with X-lite and the other FC5 with Ekiga |
| 10:40.44 | SanketMedhi | both clients register |
| 10:41.10 | SanketMedhi | amazingly, when I dial Ekiga's number, I get number incomplete |
| 10:41.43 | SanketMedhi | X-lite does not allow me to add a number like myext@myserver |
| 10:41.53 | SanketMedhi | all I can enter is the number |
| 10:42.13 | SanketMedhi | when I dial from ekiga to x-lite, I get an error on my asterisk console |
| 10:46.34 | *** join/#asterisk vgster (n=vgster@cpc2-ledn1-0-0-cust944.leed.cable.ntl.com) |
| 10:48.20 | xnon | apache is in /var/www |
| 10:48.40 | *** part/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 10:49.38 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 10:50.15 | SanketMedhi | xnon: which distro? |
| 10:50.24 | xnon | Debian Sarge |
| 10:53.19 | SanketMedhi | do you know how to use synaptic? |
| 10:54.27 | SanketMedhi | xnon: go to #debian |
| 10:54.43 | SanketMedhi | xnon: once you have LAMP working, come here |
| 10:55.02 | SanketMedhi | in fact, once you have LAMP working, go to #freepbx |
| 10:55.03 | SanketMedhi | :) |
| 10:56.42 | *** join/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 10:57.28 | EmleyMoor | Is the Logitech 350 headset OK for use with Linux-based softphones? (asking here because I will be connecting through asterisk) |
| 10:57.57 | SanketMedhi | EmleyMoor: headsets have nothing to do with Asterisk or any softphone |
| 10:58.41 | EmleyMoor | Indeed - it's more to do with support at the audio level I guess |
| 10:59.09 | SanketMedhi | if your sound system is working, it will work |
| 10:59.43 | SanketMedhi | if your system is up to date, your sound system will work :) |
| 10:59.55 | EmleyMoor | That headset is an additional sound system in its own right - so I need to check elsewhere I guess |
| 11:00.30 | SanketMedhi | EmleyMoor: by sound system I meant your software |
| 11:00.38 | SanketMedhi | just try it |
| 11:00.41 | SanketMedhi | it will work |
| 11:00.55 | SanketMedhi | if it doesn't, ask in your distribution's channel |
| 11:01.02 | SanketMedhi | ok |
| 11:01.22 | EmleyMoor | Nobody there seems to know - thanks for the advice anyway |
| 11:01.43 | SanketMedhi | np |
| 11:09.10 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 11:10.55 | Aurs | hi |
| 11:18.39 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 11:19.09 | puzzled | hi |
| 11:20.16 | Aurs | hi puzzled |
| 11:21.30 | pyrom | I can't get my sipura to register with asterisk |
| 11:21.39 | pyrom | Anything i should look for? |
| 11:22.04 | pyrom | Managed once before i upgraded the firmware, but now i can't - might've missed something but i realy dont know : Followed all guides out there. |
| 11:22.28 | pyrom | spa-3102 |
| 11:26.40 | *** join/#asterisk Dr^Mouse (n=kwagner@66.160.135.57) |
| 11:26.54 | *** join/#asterisk viperdude (n=jon@195.74.96.120) |
| 11:28.04 | Dr^Mouse | can someone please help? i am getting no effect (as shown in ztmonitor) from adjusting the zapata.conf txgain setting (on a tdm400 card, fxo module) |
| 11:28.54 | Dr^Mouse | even putting it up to stupidly high figures doesnt help. im on zaptel 1.2.7 (tried on 1.2.6 aswell) and asterisk 1.2.10 |
| 11:30.16 | Dr^Mouse | as a result i am unable to get ztmonitor to show tx level of 50%, it is always coming out as arounf 1600-1700 |
| 11:30.40 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 11:30.45 | backblue | hi* |
| 11:33.45 | *** join/#asterisk oej (n=oej@63.116.149.163) |
| 11:34.23 | Ahrimanes | hey oej :) |
| 11:34.51 | oej | Morning! |
| 11:34.56 | oej | Welcome back |
| 11:34.58 | oej | I am disconnecting soon |
| 11:35.05 | Ahrimanes | thx |
| 11:35.07 | Ahrimanes | oh no why ? |
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| 11:35.45 | Dr^Mouse | Ahrimanes - dont think he likes you :P |
| 11:35.58 | Ahrimanes | heeh |
| 11:36.06 | Ahrimanes | he knows talking to me means work ;) |
| 11:36.08 | Aurs | very soon Ahrimanes.. hehe |
| 11:36.51 | Dr^Mouse | has anyone else had problems with tdm400 cards not adjusting gain? |
| 11:37.16 | Ahrimanes | Dr^Mouse: sorry, dont use them |
| 11:37.28 | Dr^Mouse | i wish i didnt |
| 11:37.42 | Ahrimanes | hehe |
| 11:37.56 | Ahrimanes | tried calling digium ? |
| 11:37.57 | *** join/#asterisk Nivex (n=kjotte@user-0ce2nsu.cable.mindspring.com) |
| 11:38.19 | Dr^Mouse | been nothing but trouble from day one. ive become very good over the last 2 months at solving echo problems, but if i cant adjust the gain... |
| 11:38.40 | Ahrimanes | hm, annoying |
| 11:39.49 | viperdude | hi i have a zap card that is forwarding all calls to another asterisk via IAX but IAX detection is not working on the 2nd asterisk box... any ideas? |
| 11:40.05 | viperdude | oops meant DTMF detection |
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| 11:45.47 | Dr^Mouse | right, gotta go. must plug back into phone system (this customers network is virused to hell, so weve gotta keep the phones on a sepparate lan :( of meaning no remote administration either :'( ) |
| 11:45.52 | RoyK | erm |
| 11:57.56 | benjk | RoyK, I think I saw in the scrollback you wanted to know how to reject calls on PRI, did you sort it out or do you still need an answer? |
| 11:59.43 | Aurs | why should i use friend in sip.conf? or why not? |
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| 12:02.54 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
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| 12:10.41 | mut | anyone know why shell commands don't work via manager api? |
| 12:11.05 | mut | if i action: command .. command: ! touch /tmp/somefile |
| 12:11.07 | mut | it doesn't work |
| 12:11.12 | mut | nor does |
| 12:11.16 | mut | action: command .. command: ! exec touch /tmp/somefile |
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| 12:25.16 | Aurs | mut: don't know, but.. do you have permissions to do it (in manager.conf)? |
| 12:25.46 | mut | Response: Follows |
| 12:25.46 | mut | Privilege: Command |
| 12:25.46 | mut | --END COMMAND-- |
| 12:25.57 | mut | it lets me do it |
| 12:26.03 | mut | but it doesn't do anything |
| 12:26.10 | mut | if i do it at the cli |
| 12:26.11 | mut | it works |
| 12:26.22 | rg1_ | test |
| 12:27.20 | rg1_ | hey, any of you guys know if you can do a "gosub" within a While() loop, where that "gosub" contains its own While()? i.e. can you execute a while within a while? |
| 12:27.48 | benjk | what does the console say? |
| 12:27.58 | mut | same output benjk |
| 12:28.09 | mut | if i use a command that outputs data |
| 12:28.10 | mut | like |
| 12:28.12 | *** join/#asterisk fafnir (i=hahaha@unaffiliated/fafnir) |
| 12:28.13 | mut | ! ls /var/log |
| 12:28.16 | benjk | does it say something like executing foobar in new stack ? |
| 12:28.17 | mut | in the CLI it shows output |
| 12:28.24 | RoyK | benjk: it seems to be possible to reply with a PRI cause 34 |
| 12:28.29 | benjk | emphasis on "new stack" ? |
| 12:28.29 | mut | in the api it shows what i pasted above |
| 12:28.38 | RoyK | benjk: 'lying' about 'no more Bchans available' |
| 12:28.44 | rg1_ | anyone know if you can execute a While() within a While()? Can you have nested While()'s |
| 12:28.53 | benjk | if it does execute in a new stack, then you can nest |
| 12:29.10 | RoyK | rg1_: do with gotoif() instead, perhaps |
| 12:29.24 | benjk | RoyK, yes PRI_CAUSE is the one to use |
| 12:29.41 | rg1_ | thanks roy |
| 12:29.43 | RoyK | benjk: I know, and I beleive 34 is the only one that'll work... |
| 12:29.53 | benjk | if you set it to 1 and hangup, you can event fake a "this number is not in service" |
| 12:30.03 | benjk | oh really? |
| 12:30.12 | benjk | I can send anything I want |
| 12:30.17 | RoyK | but not in service will reject the call |
| 12:30.21 | benjk | including 1 |
| 12:30.25 | RoyK | 34 will say 'use the next link' |
| 12:30.31 | RoyK | or no? |
| 12:30.45 | benjk | ah, ok, I wasnt sure what your intent was |
| 12:31.13 | RoyK | my intent is to stop bothering the telco whenever i want to move a DID from one PRI link to another |
| 12:31.22 | RoyK | because they fuck up all the time |
| 12:31.23 | benjk | you can look up the PRI code in include/asterisk/causes.h |
| 12:31.51 | RoyK | http://www.quintum.com/support/xplatform/network/Q931_Disconnect_Cause_Code_List.pdf#search=%22pri%20cause%20codes%22 |
| 12:31.58 | benjk | or that :) |
| 12:32.22 | RoyK | that pdf even has some description :) |
| 12:32.33 | benjk | so you want do divert the call to another channel if it doesn't come in on the desired one |
| 12:33.53 | benjk | you may also want to look at Q.850 |
| 12:34.09 | benjk | that describes additional cause code information elements |
| 12:34.39 | caio1982 | tzafrir: looks like the patch i'm backporting to asterisk in debian trunk (t38 support for faxing) is conflicting with bristuff because that patch commands inside debian/rules. although i'm not yet sure about this, what's the best and easy way to disable bristuff so it wont get built? |
| 12:35.17 | RoyK | benjk: where can i find that? |
| 12:35.34 | benjk | I got a copy from coppice |
| 12:35.57 | RoyK | me have? |
| 12:35.57 | benjk | I can send it to you if you like |
| 12:35.57 | RoyK | roy@karlsbakk.net |
| 12:35.57 | benjk | ok |
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| 12:37.07 | caio1982 | benjk: what about sending it to caio@ueberalles.net as well? |
| 12:37.27 | benjk | sent |
| 12:37.48 | benjk | oh dear, I hope I don't have to send it to 194 people now :) |
| 12:37.55 | caio1982 | hehe |
| 12:38.24 | tzafrir | caio1982, nullify the bristuff dpatch? |
| 12:38.36 | benjk | sent |
| 12:39.22 | benjk | just run the download script without the install script |
| 12:39.43 | *** part/#asterisk negativecreep (n=xaeem@210.2.151.110) |
| 12:39.43 | caio1982 | tzafrir: i thought there was some variable for it or something like that; i'll just comment the calls to this patch then |
| 12:39.45 | benjk | that'll fetch all the stuff that bristuff fetches but not patch and not build |
| 12:39.58 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:40.24 | caio1982 | benjk: actually it's about the debian package, not the real bristuff procedures :) |
| 12:40.37 | tzafrir | caio1982, there is a var for the docs :-( |
| 12:40.40 | benjk | ah ok, dunno about that one |
| 12:41.02 | benjk | although I am running Ubuntu which is a cousin of Debian |
| 12:41.36 | benjk | RoyK, did you get the mail? |
| 12:41.39 | *** join/#asterisk iCEBrkr (i=icebrkr@69.9.167.70) |
| 12:42.01 | RoyK | benjk: yep, thanks :) |
| 12:42.26 | benjk | welcome |
| 12:45.01 | [TK]D-Fender | benjk: More like : "If a man and a woman in Arkansas get divorced.... are they still brother & sister?" |
| 12:45.29 | benjk | heh |
| 12:46.49 | mut | anyone have the latest trunk? |
| 12:46.59 | mut | test shell command via manager api? |
| 12:47.03 | mut | or even latest stable |
| 12:47.45 | benjk | are you sure you want to allow the manager api to execute shell commands? |
| 12:48.33 | mut | right now i only need it for debugging purposes on an app that interfaces with the apio |
| 12:48.34 | mut | api* |
| 12:48.55 | mut | nothing that is put in practice |
| 12:49.13 | benjk | I guess you can always use ssh |
| 12:49.28 | mut | not to test this app |
| 12:49.36 | mut | i need to be able to dump a large amount of data from the manager api |
| 12:49.40 | mut | on command |
| 12:49.48 | mut | and there is nothing that i can use to do that |
| 12:49.56 | benjk | C |
| 12:50.12 | benjk | and gcc and make |
| 12:50.15 | benjk | ;) |
| 12:50.19 | mut | heh |
| 12:51.31 | caio1982 | benjk: got your mail, thanks |
| 12:51.32 | mut | but ya know, now that i think about it |
| 12:51.43 | benjk | welcome |
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| 12:52.09 | mut | is there a free dll that does ssh? |
| 12:52.22 | benjk | DLL? |
| 12:52.24 | mut | i don't think i want to make an ssh implementation just to do what i'm thinking |
| 12:52.24 | benjk | eeeek |
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| 12:52.34 | tzafrir | mut: putty? |
| 12:52.35 | mut | website is in asp |
| 12:52.49 | mut | need to get something i can interface with it via the website |
| 12:53.14 | mut | putty? |
| 12:53.46 | tzafrir | ~google putty |
| 12:53.48 | benjk | putty is a windows ssh terminal application |
| 12:54.07 | mut | right.. |
| 12:54.18 | mut | how does that help me scripting it in asp/vbscript? |
| 12:54.26 | benjk | how would we know |
| 12:54.40 | mut | i'de assume he knew because he recommended it..? |
| 12:54.51 | benjk | this is (or at least used to be) a unix centric channel |
| 12:55.31 | benjk | or because he once got stuck trying to connect to a server via ssh when he was unfortunate enough to end up in front of a Windows box |
| 12:56.02 | mut | well |
| 12:56.07 | mut | it's not by choice i use asp |
| 12:56.23 | benjk | heh |
| 12:57.07 | tzafrir | Put a linux box and be done with it |
| 12:57.20 | tzafrir | At least then you could ask us for help ;-) |
| 12:57.35 | mut | well what i origionally asked was asterisk related |
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| 12:58.14 | rg1_ | RoyK - I did the GotoIf - here is what I got in the log. I would have thought based on this log, the logic would have jumped to the stop_asking_for_response - but it just fell through to the next statement - here is the log |
| 12:58.17 | rg1_ | <PROTECTED> |
| 12:58.27 | tzafrir | mut, I'm not sure I understand what you orginally asked |
| 12:58.41 | mut | tzafrir: you can execute shell commands via cli |
| 12:58.50 | mut | e.g ! touch /tmp/filename |
| 12:58.57 | mut | will create/touch a file into /tmp/filename |
| 12:59.00 | mut | at cli |
| 12:59.01 | tzafrir | mut, basically: no. "!" is not a manager command |
| 12:59.11 | mut | right.. it's a cli command |
| 12:59.19 | mut | which i THOUGHT thats what Action: command did |
| 12:59.22 | mut | was exec cli commands |
| 12:59.47 | tzafrir | It's implemented by the local process and not by the remote asterisk |
| 13:00.18 | mut | so it's just a dupe of all the cli commands? |
| 13:00.24 | mut | output could be totally different? |
| 13:00.27 | rg1_ | maybe RoyK is away, -- can anyone else check out that GotoIf and tell me what the heck i'm doing wrong? |
| 13:00.34 | benjk | 1 & 0 doesn't look right |
| 13:01.28 | benjk | paste the line with the GotoIf |
| 13:01.28 | *** join/#asterisk af_ (n=af@ip-173-144.sn1.eutelia.it) |
| 13:01.32 | benjk | just the one line |
| 13:01.39 | mut | tzafrir? |
| 13:02.15 | rg1_ | benjk - here it is |
| 13:02.16 | benjk | RoyK is busy reading Q.850 |
| 13:02.16 | rg1_ | exten => s,n,GotoIf([$[${TEMP_NUM_CONTINUE} < ${TQM_USER_DIALOG_MAX_ATTEMPTS}] & $[${TEMP_SPEECH_SCORE} < ${TQM_SPEECH_SCORE_MINIMUM_RESP}]]?:tqmMain_get_user_response_context,s,stop_asking_for_response) |
| 13:02.49 | rg1_ | the first condition was true; the second one was false |
| 13:03.49 | benjk | you may want to break this up |
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| 13:04.30 | rg1_ | can the condition NOT have an "&", etc? |
| 13:04.41 | benjk | and calculate the sub expressions first, then do NoOp(${intermediate-result1}, ${intermediate-result2} ...) |
| 13:04.45 | rg1_ | or do you need a single result |
| 13:04.54 | rg1_ | ah |
| 13:04.58 | rg1_ | do that with a "set"? |
| 13:04.59 | benjk | just so you see whats going on |
| 13:05.04 | benjk | poor man's debug |
| 13:05.04 | rg1_ | gotcha |
| 13:05.05 | rg1_ | thanks |
| 13:06.09 | *** join/#asterisk tecnico (n=tecnico@24.96.146.69) |
| 13:07.14 | [TK]D-Fender | rg1_: Paste the actual line from your dialplan, not its execution |
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| 13:07.32 | benjk | he already did that |
| 13:07.34 | [TK]D-Fender | rg1_: NVM... blind this morning |
| 13:07.39 | benjk | heh |
| 13:08.10 | [TK]D-Fender | rg1_: But yeah, that clearly isn't right |
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| 13:08.52 | benjk | as I said before [1 & 0] doesn't look right to me |
| 13:09.12 | [TK]D-Fender | exten => s,n,GotoIf([$[${TEMP_NUM_CONTINUE} < ${TQM_USER_DIALOG_MAX_ATTEMPTS} & ${TEMP_SPEECH_SCORE} < ${TQM_SPEECH_SCORE_MINIMUM_RESP}]?:tqmMain_get_user_response_context,s,stop_asking_for_response) |
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| 13:09.31 | [TK]D-Fender | benjk: He broke out of the first [ right away... that = bad |
| 13:11.41 | mut | k he dissapeared |
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| 13:15.05 | Jeffjohnson | i don't get a dialtone if i dial with my isdn phone. The line is still until the other end answers the phone. With my voip it works as expected. anybody have an idea? |
| 13:17.07 | puzzled | Jeffjohnson: not sure but did you try turning on early B3? |
| 13:17.25 | Jeffjohnson | puzzled: what's that? :) |
| 13:17.32 | Jeffjohnson | puzzled: and how I do it? :) |
| 13:18.09 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:19.06 | puzzled | Jeffjohnson: that tells me that you did not read the docs first :) Iirc early B3 is paasing the sounds from the telco switch. Can't help you how to do it. Read the docs... |
| 13:20.03 | Jeffjohnson | puzzled: which doc? "Asterisk: The Future of Telephony" I've searched on voip-info.org allready for a solution |
| 13:20.32 | puzzled | Jeffjohnson:the docs that are for the ISDN card you are using to hook up your isdn phone |
| 13:20.51 | Jeffjohnson | puzzled: mmh k thx :) |
| 13:20.57 | Jeffjohnson | puzzled: wait |
| 13:22.14 | Jeffjohnson | puzzled: i've forget to say that i works with sipgate.de also with my isdn phone. But if i use dusnet as provider, i have the describted behaviour. The dusnet support says me that it should also work |
| 13:22.28 | Jeffjohnson | puzzled: so i don't think that is a isdn configuration problem |
| 13:22.43 | puzzled | sorry, don't know |
| 13:26.39 | Jeffjohnson | i dont know, too :E |
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| 13:40.08 | rg1_ | Does anyone know - can you have nested GoSubs? |
| 13:40.31 | *** join/#asterisk hmmhesays (n=ohyeah@66.173.103.110) |
| 13:41.02 | hmmhesays | yup |
| 13:41.08 | rg1_ | you can? |
| 13:41.17 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:41.23 | rg1_ | how about nested While's? |
| 13:43.23 | rg1_ | hmmm ? |
| 13:45.54 | *** join/#asterisk hi365 (n=hi365@bzq-167-158.dsl.bezeqint.net) |
| 13:46.19 | RoyK | rg1_: why not just use macros? |
| 13:46.28 | hi365 | im havinh a problem with FOP: it only displays some ot the trunks. how do i go about fixing it? |
| 13:47.01 | benjk | macros or apps |
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| 13:48.31 | hi365 | im havinh a problem with FOP: it only displays some ot the trunks. how do i go about fixing it? |
| 13:49.04 | hmmhesays | rg1_: i just logged in i have no idea what you are talking about |
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| 13:52.45 | hmmhesays | i'll keep you my dirty little secret |
| 13:53.03 | caio1982 | tzafrir: is it okay or even normal the bristuff patches output lots of 'hunk #XX successed (-X offset lines)" when svn-building the trunk version of asterisk .deb? i'm afraid it's a side effect from my backported patch but i never noticed the bristuff code being applied before |
| 13:54.19 | *** join/#asterisk meppl (i=mephisto@meppl.net) |
| 13:54.34 | meppl | what does that mean? |
| 13:54.34 | meppl | Aug 21 15:53:47 ERROR[7366]: chan_zap.c:2782 zt_hangup: What is wrong with you? You cannot use cause 1 number when in state 1 |
| 13:54.52 | hmmhesays | my god meppl what is wrong with you |
| 13:55.00 | meppl | :/ |
| 13:55.17 | hmmhesays | just kidding |
| 13:55.50 | meppl | i have no idea - thats wrong with me |
| 13:56.04 | meppl | i have no clue |
| 13:56.29 | hmmhesays | is some delinquent behavior accompanied by that message |
| 13:56.56 | benjk | grep -r AST_STATE include/asterisk/*.h |
| 13:57.31 | tzafrir | caio1982, normal |
| 13:57.41 | benjk | and hangup cause 1 is "unallocated number" |
| 13:57.56 | meppl | oh okay |
| 13:58.09 | caio1982 | tzafrir: oh (sighs), good to hear :) |
| 13:58.12 | benjk | did you try something like Hangup(1) |
| 13:58.31 | tzafrir | note that I had to adapt the latest patch to 1.2.10 |
| 13:58.53 | benjk | in general, hanging up only makes sense if state is UP |
| 13:59.16 | benjk | or at least off hook |
| 13:59.26 | hmmhesays | maybe in your crazy world |
| 13:59.56 | benjk | below off hook it won't make any sense |
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| 14:15.18 | *** join/#asterisk iq (n=iq@unaffiliated/iq) |
| 14:15.21 | iq | Hi |
| 14:15.22 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 14:15.33 | profounded | hey |
| 14:15.46 | blitzrage | hypa7ia: yo! |
| 14:16.08 | profounded | i want to make sure I got this right: I get a TDM400p card and an FXS module to hook up to my phone |
| 14:16.23 | profounded | and then i get a voip line and i should be good right? |
| 14:16.23 | blitzrage | so far so good |
| 14:16.28 | blitzrage | yes |
| 14:16.31 | blitzrage | or a VoIP phone |
| 14:16.52 | profounded | cool.. and can i hook an fxs phone to a regular (not digital) phone temporaily? |
| 14:16.57 | blitzrage | there are several decent VoIP phones for approximately the same price range |
| 14:17.12 | Unistim_junky | Are there any pros/cons in choosing SER or OpenSER. Do you all value one over the other. Also, anyone know how many developers are actually assigned/working each? |
| 14:17.14 | hmmhesays | no you must use cisco or die heil CISCO |
| 14:17.14 | blitzrage | the FXS port will only work with analog phones -- non-digital |
| 14:17.41 | hmmhesays | Unistim_junky: OpenSER has more functions now |
| 14:17.45 | blitzrage | all hail CISCO! |
| 14:17.58 | profounded | what module hooks up to digital phones? |
| 14:18.06 | blitzrage | like ISDN? none |
| 14:18.22 | hmmhesays | i use openser when need be |
| 14:18.26 | blitzrage | TDM400p is an analog only card |
| 14:18.33 | profounded | maybe im confused what the difference between an analog and digital phone is |
| 14:18.43 | blitzrage | I think so |
| 14:19.01 | blitzrage | when I think digital phone, I think of something like an ISDN phone with ABCD buttons |
| 14:19.09 | blitzrage | not many people are just going to have one of those lieing around |
| 14:19.19 | profounded | most office phones are analog then right? |
| 14:19.23 | blitzrage | yes |
| 14:19.36 | hmmhesays | most office phones are a bastard child of analog and digital |
| 14:19.41 | blitzrage | that too :) |
| 14:19.46 | *** join/#asterisk daysmen3 (n=primus@host86-138-239-164.range86-138.btcentralplus.com) |
| 14:19.46 | hmmhesays | kind of like me |
| 14:19.47 | Unistim_junky | hmmhesays: I am seeing that as well. OpenSER seems to have much more indepth docs/comm |
| 14:20.03 | hmmhesays | Unistim_junky: there is a lot of documentation for SER also |
| 14:20.08 | profounded | got it.. so then repharsing my question: can i use an ordinary house phone temporaily with an FXS port.. i would assume yes? |
| 14:20.21 | blitzrage | yes -- that's what it's for |
| 14:20.33 | *** join/#asterisk breakdisk (n=breakdis@62.149.122.2) |
| 14:21.02 | benjk | you can hook up a digital phone (ISDN phone set) using a sub $50 BRI card |
| 14:21.03 | profounded | and i can upgrade later to an office phone that has extension 1, extension 2, conferance calls and all that good stuff, right? |
| 14:21.13 | blitzrage | that's what ASTERISK does |
| 14:21.13 | tzanger | benjk: if you can find a damn card in north america |
| 14:21.25 | benjk | still, you might want to consider an IP phone |
| 14:21.46 | benjk | that kinda counts as digital too |
| 14:21.46 | hmmhesays | yeah definately go the ip phone route |
| 14:21.55 | benjk | tzanger you can always order them from TW |
| 14:22.00 | Unistim_junky | hmmhesays: The manual that the quick start guide speaks of is a broken link. I was able to find SIP Express Router v0.11.0 -- Admin’s Guide. But neither SER or OpenSER is at that version yet. I will keep digging |
| 14:22.02 | blitzrage | I personally like the SPA942s, and heard decent reviews of the new Polycom IP430 |
| 14:22.13 | benjk | how many do you need, I'll but in an order again next week |
| 14:22.14 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 14:22.19 | profounded | whats the minimum you can set up ip phone for (assuming not softphone)? |
| 14:22.34 | blitzrage | min < $100, but you don't want those phones |
| 14:22.34 | hmmhesays | Unistim_junky: read the admin guide |
| 14:22.43 | blitzrage | look for aroudn $180-$200 per set |
| 14:22.48 | hmmhesays | get an spa-942 |
| 14:22.52 | [TK]D-Fender | profounded: Don't cheap out on IP phones. If you are going to spend money, do it right |
| 14:23.03 | [TK]D-Fender | SPA = bleh... no presenece, puny screens... |
| 14:23.11 | blitzrage | but that doesn't mean you have to spend $500 per phone either |
| 14:23.17 | hmmhesays | [TK]D-Fender: cheaper than the poly's |
| 14:23.22 | [TK]D-Fender | profounded: Where are you located. |
| 14:23.30 | [TK]D-Fender | hmmhesays: Mildly..... VERY. |
| 14:23.31 | profounded | alright i wont, there is no additonal hardware needed though? they hook up to ethernet? or to the server? |
| 14:23.34 | hmmhesays | the 942's have presence capability |
| 14:23.36 | profounded | i live in eastern USA |
| 14:23.41 | blitzrage | ethernet port |
| 14:23.43 | benjk | if you want to spend as little money as possible just to play around a bit and "grow Asterisk/Voip legs" then you can do that for about $50 or so |
| 14:23.49 | [TK]D-Fender | hmmhesays: I can get an IP 430 for the same price as an SPA-941 here..... |
| 14:23.51 | blitzrage | most can be powered from a switch with PoE too |
| 14:23.57 | hmmhesays | you can do that with a softphone for free |
| 14:23.58 | benjk | but if you want a more solid thing, you may want to spend 100 or more |
| 14:24.34 | profounded | ok so im cool with spending like $150 or so.. You would recommend an IP phone over a TMD400P card then? |
| 14:24.36 | [TK]D-Fender | profounded: For analog phones (a few) use ATA's like the SPA-2002. that'll give you 2 analog phones converted to SIP for $70 USD total. |
| 14:24.42 | profounded | I think it makes more sense to me |
| 14:24.57 | hmmhesays | yes |
| 14:24.59 | hmmhesays | yes |
| 14:25.01 | hmmhesays | OH YES |
| 14:25.02 | benjk | ok ip phones start at around 100 |
| 14:25.03 | [TK]D-Fender | profounded: If you WANT analog or need it then use ATA's far cheaper and flexible to deploy and use. |
| 14:25.16 | hmmhesays | ip phones are more fun |
| 14:25.18 | benjk | barbiephones start at about 50 or so |
| 14:25.22 | profounded | fuck it.. ill go IP |
| 14:25.23 | profounded | ;) |
| 14:25.37 | profounded | if i can spend 150, hook it up to my router, then why the hell not |
| 14:25.55 | hmmhesays | and it will keep you arm at night |
| 14:25.57 | hmmhesays | *warm |
| 14:26.00 | profounded | that seems to make more sense then 70 for tmd400p 70 for module and 50 for cheap phne |
| 14:26.02 | [TK]D-Fender | profounded: :Here, place for great deals on great phones : http://www.telephonydepot.com/Polycom_s/25.htm |
| 14:26.32 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 14:26.53 | hmmhesays | anyone ever deal with these netscreen routers from juniper networks? |
| 14:26.56 | profounded | thx [tk]D-Fender, thx everyone! |
| 14:26.59 | benjk | will keep you armed at night |
| 14:27.20 | hmmhesays | that too |
| 14:27.25 | benjk | need a firearm license for those IP phones in Europe though |
| 14:27.44 | mut | fukin telco! |
| 14:27.51 | mut | due to heavy calling! |
| 14:27.56 | benjk | and SIP ammo can blow up in your face |
| 14:28.04 | hmmhesays | "we've been caught with our pants down" |
| 14:28.07 | benjk | safer to use IAX ammo |
| 14:28.21 | hmmhesays | unless that ammo is bound to the wrong interface |
| 14:28.24 | benjk | but only if you shoot at a distance |
| 14:28.41 | benjk | if you shoot in your lanyard, SIP ammo is ok |
| 14:28.52 | hmmhesays | wow, we have just traveled down a dark dark path |
| 14:28.57 | hmmhesays | let us back out slowly |
| 14:29.03 | benjk | or at certified and authorised firing ranges |
| 14:30.26 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 14:30.53 | profounded | <PROTECTED> |
| 14:37.03 | yxa | is there an easy way to see the max concurrent calls on a PRI? |
| 14:37.30 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 14:37.30 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:39.02 | [TK]D-Fender | profounded: Excellent general purpose phone |
| 14:39.14 | [TK]D-Fender | profounded: PM |
| 14:41.47 | hmmhesays | bah this netscreen is a kickin my @$$ |
| 14:45.54 | *** join/#asterisk devel (n=devel@wiggum.digitalcoven.com) |
| 14:46.05 | puzzled | [TK]D-Fender: does it have a backlit screen? |
| 14:46.21 | puzzled | that's something I miss on my Cisco 7960s |
| 14:46.23 | [TK]D-Fender | puzzled: Nope, ont thin the 922 has over them. |
| 14:46.37 | puzzled | you mean linksys 942? |
| 14:46.39 | [TK]D-Fender | bleh.... can't type today. |
| 14:46.49 | puzzled | heh |
| 14:46.54 | [TK]D-Fender | puzzled: Well not sure on the 922, I know the 942 has a backlight. |
| 14:47.11 | [TK]D-Fender | 922 is really cut-rate.... |
| 14:47.32 | puzzled | don't know the 922. guess I should hit google |
| 14:47.55 | [TK]D-Fender | puzzled: And yeah I did mean 942.... |
| 14:48.25 | [TK]D-Fender | puzzled: Being in North America just forget LInksys really... Polycom is a far more solid choice. |
| 14:48.51 | puzzled | if only they add the backlit screen... |
| 14:49.07 | [TK]D-Fender | puzzled: I know, its amongst the top requests on their forums. |
| 14:49.07 | Juggie | heh, http://snakesonaplane.varitalk.com/ |
| 14:49.09 | Juggie | enjoy. |
| 14:49.29 | *** join/#asterisk crlshn (i=kvirc@operaciones3.globalnet.hn) |
| 14:49.36 | [TK]D-Fender | Juggie: I watched it last night... not bad. Far from amazing though. A shocker through & through |
| 14:49.48 | crlshn | Endpoint Question...: does Quintum Tenor AX required a especial SIP friend configuration...to allow registration. |
| 14:49.57 | puzzled | Juggie: that url doesn't do anything in FF 1.5 |
| 14:49.58 | Juggie | [TK]D-Fender, check out that site though, you can send your friend a personalized message via phone. |
| 14:50.01 | Juggie | to go see the movie. |
| 14:50.18 | Juggie | puzzled, it does for me in 1.5.0.6 |
| 14:50.19 | [TK]D-Fender | Juggie: I would never do that to a friend :) |
| 14:50.28 | Juggie | okok, its quite funny :) |
| 14:50.33 | Juggie | and its telephony related. |
| 14:50.33 | puzzled | Juggie: strange, got that one too |
| 14:50.53 | Juggie | puzzled, it uses flash8 as well |
| 14:51.18 | Juggie | the page has one image & a flash. |
| 14:52.02 | Juggie | i wonder what the backend is, could be * who knows :) |
| 14:52.13 | puzzled | Juggie: it helps if you make it "play" :) |
| 14:53.33 | Juggie | indeed. |
| 14:54.05 | vader-- | does anyone know if on the cisco 7940G or through asterisk there is a way to control the ringer types for each individual line that goes into the phone? |
| 14:54.35 | yatesy | apparently its possible in asterisk, but i've never got it to work :/ |
| 14:54.53 | *** join/#asterisk crCernier (n=crochat@adsl-84-227-76-77.adslplus.ch) |
| 14:55.22 | Juggie | see puzzled, i told you it was cool :) |
| 14:57.08 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 14:57.56 | vlt | Hello. Can you recommend a SoftPhone for connecting to asterisk running under Windows? |
| 14:58.13 | Juggie | www.xten.net |
| 14:58.14 | hmmhesays | there are many |
| 15:00.22 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 15:01.58 | *** join/#asterisk darviria (n=dvr@194-105-181-29.ifb.co.uk) |
| 15:02.02 | vlt | Juggie: Thank you. |
| 15:04.05 | *** join/#asterisk xnon (i=xnon@200.82.222.64) |
| 15:04.10 | xnon | hello |
| 15:04.24 | Anotsu | vlt: try www.sjphone.org |
| 15:04.30 | xnon | i install freepbx but i dont kno what is the username and password to administrate it |
| 15:04.40 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 15:04.40 | *** mode/#asterisk [+o mog] by ChanServ |
| 15:05.29 | Juggie | join #freepbx |
| 15:07.04 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 15:07.04 | [TK]D-Fender | xnon: : like the channel topic says.... |
| 15:07.59 | coppice | anyone familiar with alarms on Siemens EWSD switches? |
| 15:08.27 | coppice | specifically, does anyone know what "dis-sa" means? |
| 15:08.48 | *** join/#asterisk Skaag (n=hintza@212.199.180.157.static.012.net.il) |
| 15:09.19 | *** join/#asterisk Skaag (n=hintza@212.199.180.157.static.012.net.il) |
| 15:09.32 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 15:09.59 | *** join/#asterisk asterisk_bounty (n=pcm@68.159.139.234) |
| 15:10.38 | *** join/#asterisk RoyK (n=roy@ti211310a080-4327.bb.online.no) |
| 15:10.44 | GerbilWrk | Anyone have any recommendations for routing hundreds of numbers to hundreds of devices, if the devices context is a mac address, not the same number? |
| 15:11.51 | *** join/#asterisk adorah (n=Administ@87.68.145.151.cable.012.net.il) |
| 15:12.21 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.91) |
| 15:12.23 | backblue | there is any context, that asterisk executes, when it is started or reload, or something? |
| 15:13.31 | *** join/#asterisk kiddy (n=kiddy@59.93.14.194) |
| 15:13.51 | kiddy | Hi , I have configured the asterisk server and setup the extensions |
| 15:14.26 | kiddy | now I got a problem : when I make call to another extension it is not passing voice between the extenisons |
| 15:14.42 | kiddy | can anybody please tell me the reason |
| 15:14.46 | puzzled | coppice: http://www.comc.org.cn/data/bar03/bar02/bar01/2006/06/14/100001120.html |
| 15:15.08 | kiddy | The strange this is that I can hear the voice messages |
| 15:15.31 | puzzled | coppice: http://www.gzit.edu.cn/gut/magazine/xb20023/2002xb3-3/2002xb33-4.html hope your chinese is good |
| 15:19.38 | *** join/#asterisk Godsey (n=jason@pdpc/supporter/sustaining/Godsey) |
| 15:20.14 | kiddy | puzzled : Can you answer my above qn ? |
| 15:20.59 | puzzled | kiddy: nope, perhaps google the error message |
| 15:21.03 | Godsey | might someone be able to tell me a relativly stable svn revision of trunk that works w/ -addons? |
| 15:21.17 | puzzled | Godsey: don't think there is one |
| 15:21.29 | *** join/#asterisk SwK[Work] (n=SwK@64.89.118.139) |
| 15:21.29 | Godsey | I have 40773 built now and it sigsevs on load (no addons either) |
| 15:23.01 | hmmhesays | this netscreen sucks |
| 15:23.40 | kiddy | puzzled : actually I am not getting any errors , also I cannot hear anything |
| 15:25.20 | *** join/#asterisk nn (n=joseph@cdm-75-109-19-189.asbnva.dhcp.suddenlink.net) |
| 15:25.25 | nn | anyone using * with sunrocket? |
| 15:26.47 | eKo1 | kiddy: are your phones/atas and * on the same lan? |
| 15:27.02 | eKo1 | Godsey: why not use stabe? |
| 15:29.18 | coppice | puzzled: thanks |
| 15:29.32 | kiddy | eKo1 : No phones are connected through VPN |
| 15:29.35 | *** join/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 15:29.46 | TripleFFFF | how i get latest STABLE 1.2.10 ? |
| 15:29.48 | TripleFFFF | from svn |
| 15:29.54 | TripleFFFF | svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk |
| 15:29.55 | TripleFFFF | ? |
| 15:30.02 | TripleFFFF | <PROTECTED> |
| 15:30.28 | eKo1 | from branches |
| 15:30.33 | TripleFFFF | last one / |
| 15:30.36 | eKo1 | trunk has the latest devel stuff |
| 15:30.36 | TripleFFFF | <PROTECTED> |
| 15:30.37 | TripleFFFF | ? |
| 15:30.43 | TripleFFFF | i mean this one has stable crap ? |
| 15:30.46 | TripleFFFF | think i got ddoes |
| 15:30.50 | eKo1 | yeah, the last one |
| 15:30.57 | eKo1 | or go to tags/1.2.10 |
| 15:31.15 | TripleFFFF | hey |
| 15:31.18 | TripleFFFF | how lol |
| 15:31.18 | kiddy | eKo1 : you have any idea about the problem ? |
| 15:31.34 | *** join/#asterisk wwalker (n=wwalker@pdpc/supporter/sustaining/wwalker) |
| 15:32.00 | eKo1 | kiddy: well, I suspect the vpn is to blame then. |
| 15:32.10 | TripleFFFF | wats 1.2.10-netsec/ |
| 15:32.32 | eKo1 | a version with the netsec patch probably. |
| 15:32.36 | RoyK | it's the asterisk version with network security :P |
| 15:32.39 | eKo1 | TripleFFFF: you need to learn a bit about svn. |
| 15:32.40 | TripleFFFF | whats netsec is the question lol |
| 15:32.42 | TripleFFFF | ;) |
| 15:32.43 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 15:33.11 | kiddy | eKo1 : But its ringing well and hearing voice mails with good quality |
| 15:33.58 | eKo1 | kiddy: maybe the audio path from phone to phone is getting botched somewhere. |
| 15:34.21 | eKo1 | Are these sip phones/atas? |
| 15:34.47 | kiddy | yes these are sip phones |
| 15:34.56 | kiddy | Grandstream |
| 15:35.16 | coppice | puzzled: can you find me references in Hong Kong or Taiwan Chinese next time. They are easier to read :-) |
| 15:35.36 | wwalker | I'm trying to run asterisk on openwrt. It comes up and registers with my main asterisk server. sjphone registers with it. then it dies in the next 15 seconds. no errors in any log (full is on, verbose 30, sip debug, debug 30). Any pointers as to what to try next? |
| 15:36.11 | TripleFFFF | <PROTECTED> |
| 15:36.13 | TripleFFFF | hmmm |
| 15:36.19 | TripleFFFF | ~bugs zonelock |
| 15:36.23 | TripleFFFF | shit |
| 15:36.42 | *** join/#asterisk roving_prole (n=Harper@72-254-127-241.client.stsn.net) |
| 15:36.49 | RoyK | ~lart TripleFFFF |
| 15:36.56 | TripleFFFF | yeah |
| 15:37.22 | TripleFFFF | weird.. centos. zaptel.. from tags.. 1.2.7 |
| 15:37.44 | RoyK | centos is evil |
| 15:38.18 | TripleFFFF | well you guys change your imind ever other fucking week |
| 15:38.26 | wwalker | TripleFFFF's box is now completely broken |
| 15:38.27 | TripleFFFF | you guys said to use centos.. wich i did |
| 15:38.44 | RoyK | wwalker: slackware 2.1 is from '94 |
| 15:38.48 | *** join/#asterisk twisla (i=twisla@lutin.jard.in) |
| 15:39.02 | jbroome | rule #1 of IRC: Everyone is full of shit |
| 15:39.06 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:39.07 | RoyK | erm |
| 15:39.09 | RoyK | i meant 3.2 |
| 15:39.12 | jets | now now lets play nice in the sandbox this morning! |
| 15:39.14 | RoyK | that one came with linux 1.1.59 |
| 15:39.17 | wwalker | RoyK: true. How does that differ from today's slackware? |
| 15:39.24 | RoyK | wwalker: a little :P |
| 15:39.46 | TripleFFFF | aitn that the src crap ? |
| 15:39.50 | TripleFFFF | ~bugs zaptel |
| 15:39.54 | wwalker | I started with 53 floppies (was it 53??) and loved slack. |
| 15:39.59 | TripleFFFF | where we had to sed the src |
| 15:40.06 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:40.12 | wwalker | it just hasn't moved forward in 10 years |
| 15:41.05 | RoyK | ~bugs |
| 15:41.18 | TripleFFFF | ~bugs |
| 15:41.27 | RoyK | ~bugs jbot |
| 15:41.35 | RoyK | ~lart himself |
| 15:41.54 | macTijn | ehehe |
| 15:42.09 | RoyK | try ~lart jbot :P |
| 15:42.10 | macTijn | that must hurt :) |
| 15:42.16 | trelane_ | jbot ab00ze! |
| 15:42.18 | macTijn | ~lart jbot |
| 15:42.18 | *** join/#asterisk sb_mx (n=sb_mx@200.78.229.18) |
| 15:42.22 | macTijn | grin |
| 15:42.24 | trelane_ | haha |
| 15:42.37 | macTijn | RoyK: uhuh, same here |
| 15:42.40 | macTijn | hut what about MFM, ESDI |
| 15:42.46 | RoyK | sure |
| 15:42.48 | TripleFFFF | ~centos |
| 15:42.50 | jbot | i guess centos is better than Fedora Core except for that silly bug, see ~centosbug for details |
| 15:42.50 | macTijn | s/hut/but/ |
| 15:42.50 | RoyK | MFM was nasty |
| 15:42.54 | TripleFFFF | ~centosbug |
| 15:42.56 | jbot | hmm... centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 15:43.21 | RoyK | tuning the disk interleave down to perhaps five or four? |
| 15:43.33 | macTijn | 4 was perfect for Seagates |
| 15:43.37 | RoyK | hehe |
| 15:43.48 | RoyK | speeding it up, closing into 300kB/s |
| 15:43.52 | macTijn | I had 6 on a full height toshiba ESDI |
| 15:43.55 | macTijn | yeah |
| 15:44.03 | macTijn | that stuff was *fast* |
| 15:44.03 | macTijn | ;) |
| 15:44.07 | RoyK | :) |
| 15:44.14 | TripleFFFF | wow |
| 15:44.15 | TripleFFFF | ed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h |
| 15:44.18 | TripleFFFF | is not right |
| 15:44.26 | macTijn | try the s in front of it |
| 15:44.28 | TripleFFFF | that pulls.. /2.6.9-34.0.2.ELsmp-i686 |
| 15:44.35 | RoyK | TripleFFFF: sed, perhaps..... |
| 15:44.39 | TripleFFFF | when it should pull 2.6.9-34.0.2.EL-smp-i686 |
| 15:44.43 | TripleFFFF | missing -dash |
| 15:44.50 | macTijn | <- gone, drinking beer |
| 15:44.52 | TripleFFFF | sed i know lol pasted part |
| 15:44.56 | TripleFFFF | not the point |
| 15:45.05 | kiddy | eKo1 : I found the problem and solved the issue |
| 15:46.00 | kiddy | eKo1 : Actually if you want to connect extensions through VPN then you have to allow the ports 10000:20000 in your firewall |
| 15:46.07 | *** join/#asterisk muppetmaster (n=jasongoe@169.red-81-184-73.user.auna.net) |
| 15:46.08 | kiddy | I done this and get it work |
| 15:46.15 | muppetmaster | Hello all |
| 15:46.25 | TripleFFFF | actually i got to patch on each frigin make ? |
| 15:46.27 | TripleFFFF | weid |
| 15:46.45 | kiddy | One question : Is there is any other way than VPN for secure VOIP ? |
| 15:46.51 | muppetmaster | In the SVN TRUNK of Asterisk, the latest from today, has something changed in interacting with the AGI in order to obtain variables. On v1.2.10 (I am using the Ruby RAGI lib) I do a "GET VARIABLE EXTEN" and all works fine. |
| 15:46.55 | *** part/#asterisk nn (n=joseph@cdm-75-109-19-189.asbnva.dhcp.suddenlink.net) |
| 15:46.58 | muppetmaster | In SVN TRUNK I do it and I get a NULL back. |
| 15:47.11 | muppetmaster | Same code set, and this was working as of a couple of days ago on SVN TRUNK. |
| 15:47.16 | muppetmaster | Anyone have any ideas? |
| 15:47.37 | *** join/#asterisk profounded (n=pro@ool-44c4eae2.dyn.optonline.net) |
| 15:47.58 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 15:48.33 | nextime | what exactly are "nbsd and nbsclient"? i can't found any information on google other than some cvs files from digium |
| 15:49.18 | TripleFFFF | is that the DDOS ? Aug 21 06:49:34 DEBUG[9250] chan_iax2.c: Immediately destroying 4, having received INVAL |
| 15:49.30 | TripleFFFF | i got like 40 of these then a dos |
| 15:49.47 | *** join/#asterisk jtodd (n=jtodd@adsl-75-24-91-221.dsl.pltn13.sbcglobal.net) |
| 15:51.40 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:51.40 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:52.08 | RoyK | TripleFFFF: they're coming for you |
| 15:52.16 | TripleFFFF | yeah it seems |
| 15:52.24 | TripleFFFF | so INVAL is the bug indeed ? |
| 15:52.27 | TripleFFFF | or im paranoid |
| 15:52.28 | RoyK | better hide |
| 15:52.29 | RoyK | run |
| 15:52.34 | RoyK | turn off all computers |
| 15:52.40 | TripleFFFF | hehee RoyK |
| 15:52.50 | *** join/#asterisk anonymouz666 (n=anonymou@h1e2.compuland.com.br) |
| 15:56.11 | crCernier | I'm searching for PA1688 firmware source code ! Anybody knows something about that ? |
| 15:57.32 | *** join/#asterisk uk-wombat (n=root@82.163.6.212) |
| 15:58.44 | *** join/#asterisk De_Mon (n=de_mon@fl-69-69-153-77.dyn.embarqhsd.net) |
| 15:59.40 | *** join/#asterisk I-MOD (n=opticron@gateway.digium.com) |
| 16:01.29 | Godsey | I fixed my SIGSEGV |
| 16:01.42 | *** join/#asterisk eKo1 (n=eKo1@190.4.7.90) |
| 16:01.50 | Godsey | for some reason it resolved itself when I rm'd svn tree and checked out again |
| 16:02.01 | Godsey | svn up didn't do something correctly I guess :) |
| 16:02.11 | mog | which version of asterisk are you running TripleFFFF ? |
| 16:02.47 | file | mog: yay loader |
| 16:02.54 | mog | i know |
| 16:03.25 | mog | im building my single binary |
| 16:03.39 | TripleFFFF | 1.2.9.1 |
| 16:03.42 | TripleFFFF | now upped to 1.2.10 |
| 16:03.55 | TripleFFFF | hey mog whats up |
| 16:04.11 | mog | hey TripleFFFF you know your name has 4F |
| 16:04.24 | TripleFFFF | yeah |
| 16:05.15 | coppice | crCernier: try http://www.aredfox.com/ |
| 16:07.05 | *** join/#asterisk Qb3rt (n=jhgjkgui@58.68.252.216.dsl1.colba.net) |
| 16:09.56 | vader-- | hmmm this sucks |
| 16:10.22 | vader-- | i have cisco 7940G phones using the sip firmware with asterisk and i have two lines going to some phones and i want to set a different ring for each line and i can't seem to do it |
| 16:10.46 | *** join/#asterisk clyrrad (n=ddd@CPE001195f553c7-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 16:13.41 | *** join/#asterisk mishehu (i=mishehu@cshells.shavedgoats.net) |
| 16:14.17 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 16:16.07 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 16:18.17 | eKo1 | vader--: cisco == nightmare |
| 16:18.29 | Qwell | no, cisco with sip firmware == nightmare |
| 16:19.21 | Juggie | cisco sucks :) |
| 16:20.52 | *** join/#asterisk moon06 (n=michael@cim06-1-82-228-240-97.fbx.proxad.net) |
| 16:21.07 | *** join/#asterisk ToTo (n=ToTo@host212-207.pool872.interbusiness.it) |
| 16:23.04 | anonymouz666 | J Rosenberg works for CISCO, they firmware should be at least good heh |
| 16:23.07 | *** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.91) |
| 16:23.21 | *** join/#asterisk Dr-Linux|work (n=Linux@202.125.139.198) |
| 16:23.31 | Dr-Linux|work | hi guys .. |
| 16:24.03 | Dr-Linux|work | i want all /var data should go to /u drive, |
| 16:24.24 | Dr-Linux|work | should i go this way or i need to do re-installation of server |
| 16:25.40 | eKo1 | err, what does this have to do with *? |
| 16:25.49 | *** join/#asterisk joburg (n=voipmagi@vc-196-207-38-156.3g.vodacom.co.za) |
| 16:26.50 | wwalker | anyone here built a recent version of asterisk for OpenWRT? |
| 16:27.47 | Dr-Linux|work | eKo1, aww, /var is an important partition for as asterisk .. isn't it? |
| 16:28.24 | Dr-Linux|work | eKo1, and /u is useless for asterisk .. that is already created by my client .. |
| 16:29.07 | Dr-Linux|work | and he doesn't have /var partition so that will take place in / which is only 3 GB |
| 16:29.09 | eKo1 | Dr-Linux|work: not if you mod. the Makefile to use it. |
| 16:29.34 | Dr-Linux|work | eKo1, actually it will be hight production IVR server, |
| 16:30.07 | Dr-Linux|work | eKo1, what you should, should i suggest them to re-install the machine with required partitions for asterisk. |
| 16:30.11 | Dr-Linux|work | ? |
| 16:31.02 | eKo1 | Well, first of /var is a dir., not a partition. |
| 16:31.36 | Dr-Linux|work | eKo1, you are right in current case |
| 16:31.36 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 16:31.48 | eKo1 | You can mount /var on its own partition though. |
| 16:31.59 | Dr-Linux|work | eKo1, but we create /var partition during installation , not a directory |
| 16:32.38 | Dr-Linux|work | while he didn't create this partition, so it's not it's a dir |
| 16:32.39 | *** join/#asterisk soylentgreen (n=fgast@193.238.89.34) |
| 16:33.39 | eKo1 | I'm having a really hard time understanding you Dr-Linux|work |
| 16:33.51 | vader-- | blah seems like no one has figured out how to do this |
| 16:33.51 | Dr-Linux|work | anyway .. it will not take much time to him to re-install the server, so later on if someone is else uses hands in this server, he/she she should easily |
| 16:33.54 | mut | wonderful |
| 16:34.02 | mut | i spent half my day dealing with a telco outtage |
| 16:34.04 | Dr-Linux|work | eKo1, it's okey thanks for your help |
| 16:34.10 | mut | now it's finally 'fixed' |
| 16:34.10 | vader-- | you can make the default ring on the cisco phones ring in a different pattern but it's sooo small of a change in the ring it's almost unnoticable |
| 16:34.20 | mut | other than one of my t1 are dead |
| 16:34.40 | *** join/#asterisk SuPrSluG (n=SuPrSluG@pool-71-243-177-207.bflony.east.verizon.net) |
| 16:34.49 | SuPrSluG | hello all |
| 16:35.15 | eKo1 | mut: your telco. had an outage? |
| 16:35.24 | mut | heh |
| 16:35.37 | mut | yea |
| 16:35.48 | mut | sporatic crap going on in the lower penninsula |
| 16:35.56 | mut | and in the upper it was pretty much all dead |
| 16:35.57 | *** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it) |
| 16:36.01 | eKo1 | I wouldn't worry. If you can't call, then noone can. |
| 16:36.08 | mut | oh but i can |
| 16:36.08 | SuPrSluG | i'm having dtmf issues. 12345 is recognized as 123445 |
| 16:36.10 | mut | and they can |
| 16:36.23 | mut | all of our on network (voip and pots) customers can still call us |
| 16:36.23 | eKo1 | That isn't an outage then. |
| 16:36.33 | mut | but they can't call outside the network |
| 16:36.43 | mut | and people can very very very randomly call in |
| 16:36.46 | mut | that is an outage |
| 16:36.58 | eKo1 | An outage is when nothing works. |
| 16:37.07 | mut | so what is this? |
| 16:37.09 | eKo1 | I call that routing problems. |
| 16:37.22 | mut | was actually signalling issues |
| 16:38.00 | eKo1 | Like when your NSP fucks up their BGP settings and you're disconnected from half the Internet. |
| 16:38.12 | BladeRunner05 | hola all, using automon => *1 in features.conf and exten => s,1,Set(DYNAMIC_FEATURES=automon) exten => s,2,Dial(SIP/xxxxx,,wW) I can record the call pressing *1. I need to record all call automatically, how can I do that ? |
| 16:38.15 | eKo1 | That is a routing problem, not an outage. |
| 16:38.20 | mut | eh well i call it an outage |
| 16:38.32 | mut | if customers can't call out |
| 16:38.35 | mut | then its an outage |
| 16:39.10 | eKo1 | I'm having one-way audio with my outbound calls... |
| 16:39.22 | eKo1 | Is that an outage? |
| 16:39.30 | mut | yes |
| 16:39.42 | Assid | is there such a thing as secured sip calls? |
| 16:39.43 | eKo1 | Oh well... |
| 16:40.55 | mut | did something screw up on the net today with some major routers? |
| 16:41.07 | mut | cause i had netsplits on like 4 different networks all approx the same time |
| 16:41.59 | eKo1 | In NA, Europe or? |
| 16:42.06 | mut | na |
| 16:43.10 | mut | bbfew |
| 16:43.29 | *** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net) |
| 16:43.58 | joburg | BladeRunner05 : use the Monitor() app before Dial in your dialplan |
| 16:44.08 | *** join/#asterisk Givemelove (n=non@208.57.229.162) |
| 16:45.07 | *** join/#asterisk heka (n=cingerr@80.80.175.130) |
| 16:46.47 | Zodiacal | anyone remapped a polycom's hard key's before? how would i change say the hard transfer key to a two digit sequence. i.e. *3? i tried adding this line to my sip.cfg key.IP_600.37.function.prim="DialpadStar" but it doesn't seem to even change it to the *. is this a bug with the phones firmware maybE? |
| 16:48.14 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 16:48.50 | *** part/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 16:49.02 | *** join/#asterisk L-info (n=linfo@62.69.102.99) |
| 16:50.36 | *** join/#asterisk FuriousGeorge (n=FuriousG@ool-43536ea8.dyn.optonline.net) |
| 16:51.50 | *** join/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 16:53.29 | blitzrage | ~centos |
| 16:53.30 | jbot | i heard centos is better than Fedora Core except for that silly bug, see ~centosbug for details |
| 16:53.36 | blitzrage | ~centosbug |
| 16:53.39 | jbot | i guess centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 16:54.35 | blitzrage | handy |
| 16:54.41 | trelane_ | yepyep |
| 16:54.49 | *** join/#asterisk hfb (n=hfb@pool-71-106-220-165.lsanca.dsl-w.verizon.net) |
| 16:54.59 | joburg | why do you think centos would be better - seeing that it has bugs? |
| 16:55.14 | blitzrage | everything has bugs |
| 16:55.23 | blitzrage | you just learn where they are and work around them |
| 16:55.40 | jbroome | or god forbid, fix them |
| 16:55.58 | blitzrage | or that |
| 16:56.10 | wwalker | jbroome: one man's bug is another man's feature |
| 16:56.25 | [TK]D-Fender | Zodiacal: What SIP rev? |
| 16:56.38 | eKo1 | canreinvite is killing me |
| 16:56.43 | wwalker | more than half of the bugs I report, on OSS or commercial apps, come back as "we like it broken that way" |
| 16:56.47 | Zodiacal | tkd-fender 1.6.6.2 i think i will have to double check 1 sec |
| 16:57.18 | joburg | is blitz really leif ? |
| 16:57.22 | [TK]D-Fender | Zodiacal: I remapped the DND to Messages on a 301 on 1 system, and attempting to do it for another failed mysteriously. |
| 16:57.44 | Zodiacal | 1.6.6.0036 exactly |
| 16:57.45 | blitzrage | yes |
| 16:57.48 | blitzrage | I really am me |
| 16:58.06 | [TK]D-Fender | joburg: Leif is Leif! NA na na na na! |
| 16:58.12 | *** join/#asterisk eNEMY^x (n=eqwrweqr@c213-158-248-202.static.sdsl.no) |
| 16:58.49 | [TK]D-Fender | blitzrage: You can't be "me", I"m ME! You're YOU! Get your head on straight! |
| 16:58.54 | joburg | blitz : whats really happening at astricon as far as steve&olaf is concerned ? |
| 16:59.03 | eNEMY^x | When I use $AGI->stream_file within a script, in some routines, I have to execute the cmd twice to actually get asterisk to play it... anyone know of this? is it a bug? |
| 16:59.04 | blitzrage | who's olaf? :) |
| 16:59.04 | [TK]D-Fender | blitzrage: And all I want is........ |
| 16:59.09 | blitzrage | ! ! ! |
| 16:59.20 | Zodiacal | [tk]d-fender what version are you using? |
| 16:59.26 | joburg | the johansen guy |
| 16:59.36 | [TK]D-Fender | That'd Be Olle |
| 16:59.41 | joburg | soory i men olle |
| 16:59.43 | blitzrage | you mean Olle -- and he won't be at Astricon |
| 16:59.53 | joburg | got a few nasty emails from him concerning steve.... |
| 16:59.54 | [TK]D-Fender | Zodiacal: On the working one, 1.6.6 as well, the failing one is 1.6.7 |
| 17:00.14 | joburg | siad steve owes him money etc etc |
| 17:00.23 | blitzrage | yah well, people can say all sorts of things |
| 17:00.28 | blitzrage | doesn't make them true |
| 17:00.38 | joburg | sounded like astricon was falling apart .... |
| 17:00.48 | blitzrage | nope -- we just sold out our exhibition hall |
| 17:00.54 | blitzrage | so I'd say it's far from failing |
| 17:01.10 | blitzrage | and we still have 2 months+ to go |
| 17:01.11 | joburg | hmmm |
| 17:01.20 | Zodiacal | [TK]D-fender and your syntax is simmiler to mine? on the working one? |
| 17:01.38 | Zodiacal | in the <keys> tag of course.. |
| 17:01.45 | Givemelove | Guys |
| 17:01.59 | Givemelove | I do have a problem with my asterisk platform |
| 17:02.09 | Givemelove | since friday, I have no audio at all |
| 17:02.15 | [TK]D-Fender | Zodiacal: Yes, quite. **HOWEVER*** please note that there are discrepencies betweent he PHOTO shoing the key numbering, and the TABLE describing them. |
| 17:02.34 | Givemelove | I tried to revert to old versions of my config files |
| 17:02.36 | Givemelove | no luck |
| 17:02.40 | [TK]D-Fender | Zodiacal: So far the photo appears to be more accurate. |
| 17:02.41 | joburg | givemelove : you ears ok ? |
| 17:03.17 | Zodiacal | tkd-fender interesting.. i used the table but maybe its wrong for my key aswell. i'll try the DND key like you did |
| 17:03.32 | joburg | did you change your conf files on friday? |
| 17:03.37 | Zodiacal | however, it does remap it.. just not to the char i specified |
| 17:03.44 | Zodiacal | some times it adjusts the volume :P |
| 17:03.54 | [TK]D-Fender | Zodiacal: Ok thats not good... |
| 17:04.16 | [TK]D-Fender | Zodiacal: SOMETIMES?! Inconsistant = VERY bad |
| 17:04.21 | Zodiacal | well, allways.. |
| 17:04.40 | Givemelove | joburg : which means? |
| 17:04.40 | Zodiacal | when i used a differnt syntax to map to a speeddial |
| 17:04.41 | Zodiacal | but the speed dial was *3 |
| 17:04.43 | Zodiacal | it *allways* ups the volume |
| 17:04.44 | Zodiacal | very strange |
| 17:05.18 | Zodiacal | tkd-fender i wanted to map to a hard key because i want it to dial the char's during the call insted of placeing the current call on hold like the speed dial keys do.. do you know if this is what i need? |
| 17:05.23 | *** join/#asterisk TrixVox (n=Spy007@c-69-248-121-104.hsd1.nj.comcast.net) |
| 17:05.25 | Zodiacal | or will work for what i need rather |
| 17:05.47 | [TK]D-Fender | Zodiacal: And unfortunately you can't do what you're looking to do. You can't assign a speed-dial to and existing channel, it will always attempt to open a new one. What are you trying to do by DTMF again? |
| 17:06.08 | Zodiacal | place a call on park with one key press |
| 17:06.26 | Zodiacal | i have the code in * to do it, i can assign it to any key or key sequence and it works if i dial those keys during a call |
| 17:06.31 | [TK]D-Fender | zod, sorry... 2 buttons required minimum. |
| 17:06.58 | Zodiacal | i just wanta map them to a hard key :( |
| 17:07.02 | [TK]D-Fender | Zodiacal: Mind you Polys do have some sort of Parking functionality but its poorly documented. |
| 17:07.36 | Zodiacal | maybe i can make it like ** then |
| 17:07.40 | Zodiacal | i guess thats the best i can do. |
| 17:08.50 | *** join/#asterisk willy1234 (n=IceChat7@62.231.36.194) |
| 17:09.18 | willy1234 | is it east to install a TDM400 |
| 17:09.24 | willy1234 | easy |
| 17:09.31 | crlshn | how do i enable the G729 codec in the TrixBox asterisk distribution ? |
| 17:09.33 | trelane_ | willy1234, simple, stick it in the machine and call digium ;) |
| 17:09.36 | FuriousGeorge | ive been using asterisk 1.2.9.1 and i gotta restart asterisk like every day, or my users start saying things like "the phones are going haywire". im about to install another server and i'm a little scared :) |
| 17:09.45 | *** part/#asterisk mtaht4 (n=m@c-71-198-23-124.hsd1.ca.comcast.net) |
| 17:09.53 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 17:09.54 | joburg | willy : extremely easy |
| 17:09.55 | trelane_ | crlshn, step 1: read the channel topic, step 2: join #freepbx, step 3: ask there :) |
| 17:10.06 | carrar | FuriousGeorge, best not to install the bleeding edge release |
| 17:10.09 | crlshn | donne |
| 17:10.14 | crlshn | no reply yet |
| 17:10.25 | joburg | crlshn : have you paid the $10 ? |
| 17:10.45 | jbroome | crlshn: make sure you keep us updated |
| 17:10.48 | crlshn | how ? where? |
| 17:11.05 | [TK]D-Fender | Zodiacal: nope, still not viable. You can't "mcaro" it, and you can't make it send on the existing channel...... |
| 17:11.30 | Zodiacal | tkd-fender i mean ill just have the user press ** to park |
| 17:11.49 | [TK]D-Fender | trelane_: My Sangoma S518 ADSL card works rather beautifully :) |
| 17:11.49 | Zodiacal | i can't think of a better way. i tried researching how polycoms park and they don't seem to work that great with asterisk park feature |
| 17:11.56 | joburg | crlshn : the g729 codec cost $10 / channel |
| 17:12.01 | Zodiacal | this patch i installed seems to work ok |
| 17:12.08 | trelane_ | [TK]D-Fender, right but I want to support digium, though I'm drooling over just that card :) |
| 17:12.15 | [TK]D-Fender | Zodiacal: Same thing really. I use app_valetparking... vastly superior..... |
| 17:12.33 | Zodiacal | tkd-fender i just wish polycoms speed dials worked like cisco's where they dial during the call |
| 17:12.34 | joburg | Sangoma works beutifully |
| 17:12.44 | Zodiacal | but nooo they have to put the call on hold.. sorry venting.. |
| 17:12.49 | [TK]D-Fender | Zodiacal: Yeah, add it to the wish-list. |
| 17:12.58 | Zodiacal | is there a polycom wish list? |
| 17:13.01 | [TK]D-Fender | Zodiacal: Trust me if I had more sway I'd use it... |
| 17:13.13 | [TK]D-Fender | Zodiacal: the y have a feature request forum on their site |
| 17:13.37 | Zodiacal | okie i might as well vote i'll do it now thanks tho! |
| 17:13.41 | [TK]D-Fender | Zodiacal: as it is they are the best thing going, so I'm pretty happy as-is |
| 17:13.55 | [TK]D-Fender | Zodiacal: Vote for it to be FULLY scriptable. |
| 17:14.16 | Zodiacal | yeah i might as well say i want the real parking features via softkeys etc |
| 17:15.12 | FuriousGeorge | carrar: bleeding edge? im using 1.2.9.1 1.2.10 is the newest release |
| 17:15.24 | [TK]D-Fender | Zodiacal: so you can do thinkgs like <macro name=x useablewhen=incall><transfer/><blind/><senddtmf dtmf=1234/></macro> |
| 17:16.07 | [TK]D-Fender | Zodiacal: So you can do extended sequences. Very viable to to it several comprehensive ways. |
| 17:16.16 | *** join/#asterisk uk-wombat (n=root@82.163.6.212) |
| 17:16.20 | [TK]D-Fender | Zodiacal: Could actually be even easier. |
| 17:16.29 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 17:17.06 | Assid | err.. if 1 extension calls another extension on 729 and if both support 729, that wouldnt need a codec would it? |
| 17:17.54 | joburg | yes it would ned a codec - the g729 codec |
| 17:17.54 | [TK]D-Fender | Assid: Nope. As long as * never needs to inject audio |
| 17:17.55 | *** part/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 17:18.32 | willy1234 | so u put the card in and rebuild zaptel |
| 17:18.39 | hmmhesays | can you help me as i'm starting to burn all alone |
| 17:18.41 | willy1234 | is that it? |
| 17:18.44 | joburg | g729 : need no license for pass-through |
| 17:18.51 | Assid | right |
| 17:18.59 | Assid | so only if it went to voicemail |
| 17:19.26 | trelane_ | [TK]D-Fender, I'm assuming I can get similar performance out of a pentium 4 to what I've been used to with my cisco 3725 on a DSL line (my voip still goes through even when getting DDoS'd) |
| 17:19.58 | Assid | one sec.. rebooting my router |
| 17:20.28 | joburg | willy : zaptel and zapata.conf ! |
| 17:20.28 | trelane_ | [TK]D-Fender, ordered an S518 |
| 17:20.28 | [TK]D-Fender | trelane_: Yeah since its a raw device w/o ehternet buffering etc you have much better BW control, etc. |
| 17:20.30 | hmmhesays | so how many concurrent calls do ya'll think I can get through openser handling the media |
| 17:20.42 | *** part/#asterisk willy1234 (n=IceChat7@62.231.36.194) |
| 17:20.45 | trelane_ | [TK]D-Fender, well I just turned all the caching off on my WIC-1ADSL |
| 17:20.49 | [TK]D-Fender | trelane_: I never got to actually tweaking anything like that and jsut got it to reduce the wiring zoo on my box. |
| 17:20.53 | *** join/#asterisk Assid (i=assid@203.115.83.215) |
| 17:21.12 | [TK]D-Fender | hmmhesays: On RS-232 not much... ;) |
| 17:21.22 | Zodiacal | tkd-fender know where the polycom forum is off hand? i can't seem to find it :/ |
| 17:21.25 | trelane_ | [TK]D-Fender, I run a blacklist and therefore idiots who think that their e-mail not going through is grounds to commit a felony cause occasional problems. |
| 17:21.34 | [TK]D-Fender | Zodiacal: no, not offhand.... |
| 17:22.27 | hmmhesays | [TK]D-Fender: haha |
| 17:22.39 | hmmhesays | say I have 10 meg up and down |
| 17:22.53 | Assid | damn.. i wish VoicePulse supported 729 |
| 17:22.53 | trelane_ | [TK]D-Fender, you know I occasionally consider starting a company providing optical class (OC-X) PCI-Express cards |
| 17:23.00 | hmmhesays | i can't find many benchmarks for openser |
| 17:23.48 | *** join/#asterisk evisu (i=hIRC@bzq-88-154-45-231.red.bezeqint.net) |
| 17:23.59 | *** join/#asterisk finejava (n=12345@60.50.245.204) |
| 17:24.24 | finejava | hi |
| 17:24.27 | finejava | question |
| 17:24.31 | trelane_ | [TK]D-Fender, I figure my application will really put wanpipe/sangoma through it's paces |
| 17:24.35 | trelane_ | finejava, yo |
| 17:24.40 | trelane_ | answer |
| 17:24.41 | joburg | hi |
| 17:24.42 | trelane_ | fin |
| 17:24.50 | trelane_ | (ghetto tcp/ip) |
| 17:24.56 | finejava | i've purchase and installed ABE A1.5 |
| 17:25.09 | finejava | can i install the B1.1 for free |
| 17:25.10 | *** join/#asterisk Dovid (n=dovi5988@pool-71-250-15-227.nwrknj.east.verizon.net) |
| 17:25.18 | finejava | or do i need to purchase a license upgrade? |
| 17:25.42 | Qwell | finejava: I would suggest calling Digium sales, and asking them. |
| 17:25.43 | trelane_ | finejava, were I you, I'd use the support you paid for and call digium and ask them, I (and most here) don't work for digium, they'd be the ones with the authoritive answer |
| 17:25.47 | Qwell | They would know for sure. |
| 17:25.52 | trelane_ | Qwell, damn you for beating me! |
| 17:25.55 | trelane_ | :) |
| 17:25.56 | Qwell | I do work for Digium, and I don't know. :p |
| 17:26.07 | TripleFFFF | <PROTECTED> |
| 17:26.07 | *** join/#asterisk zoa (n=d@pirus.securax.be) |
| 17:26.08 | zoa | hey ho |
| 17:26.12 | TripleFFFF | whats this it garbled my line |
| 17:26.13 | Qwell | zoa: y0 |
| 17:26.29 | [TK]D-Fender | trelane_: well they've been in the business for like 20 years so odds are it'll mathch what one should expect from them. |
| 17:26.34 | *** join/#asterisk intralanman (n=lanman@pool-72-82-74-171.nrflva.east.verizon.net) |
| 17:26.38 | finejava | ehmm...let me write a short email to them |
| 17:26.40 | TripleFFFF | saw like 10000 on cli |
| 17:27.36 | finejava | anyway...does any 1 here know whether the dell 1850 is compatible with the te412p? |
| 17:28.11 | finejava | the hp dl140 is not compatible with the te412/410p |
| 17:28.36 | finejava | the card is sharing the irq with both integrated eth |
| 17:28.41 | crlshAWAY | <PROTECTED> |
| 17:28.46 | crlshn | sorry |
| 17:28.58 | crlshn | i had 2 CPE with |
| 17:29.05 | crlshn | g729 licenced |
| 17:29.17 | crlshn | how do I enable passthru ? |
| 17:29.41 | Qwell | You don't need to do passthrough if you have licenses |
| 17:29.48 | crlshn | if i dial sip:23@192.168.112.2:5061 from my Eyebeam the call connects |
| 17:30.09 | FuriousGeorge | so this polarity reversal, should i worry about that? its never been a problem |
| 17:30.32 | FuriousGeorge | zt_handle_event: Ring/Off-hook in strange state 6 on channel 5 |
| 17:30.34 | crlshn | but if i dial thru the *....the calls says 603 declinne |
| 17:30.48 | crlshn | if i dial sip:23@192.168.112.2:5061 from my Eyebeam the call connects |
| 17:30.50 | crlshn | but if i dial thru the *....the calls says 603 declinne |
| 17:30.54 | FuriousGeorge | ive never found a good source online for further explanaition of some of these error codes |
| 17:31.16 | eKo1 | check the code |
| 17:31.34 | FuriousGeorge | eKo1: i mean for the 99.9% of the population who dont write code |
| 17:31.59 | joburg | see the console for details : hte console never lies |
| 17:32.00 | eKo1 | more like 99.9999% |
| 17:32.11 | FuriousGeorge | eKo1: i know some C, but i'm not gonna understand the code that goes into asterisk |
| 17:32.19 | eKo1 | why not? |
| 17:32.25 | evisu | and to prove that 99.999 statistic.... whats the command to wait 10 seconds before dialing out a trunk? exten => s,1,Wait(10) doesnt seem to work out too well.. |
| 17:32.41 | Qwell | evisu: Wait() works |
| 17:32.45 | eKo1 | evisu: why not? |
| 17:32.54 | Qwell | What are you trying to do exactly? |
| 17:32.55 | evisu | it just waits and then hangs up |
| 17:33.04 | Qwell | well, do you have anything after priority 1? |
| 17:33.21 | FuriousGeorge | eKo1: what do you mean why not? there is a difference between "i can write some c code" and understanding how chan_zap code works |
| 17:33.24 | evisu | exten => s,1,Wait(10) |
| 17:33.24 | evisu | exten => _011.,1,Macro(dialout-trunk,3,${EXTEN},,) |
| 17:33.55 | Qwell | evisu: It doesn't work like that :) |
| 17:33.56 | eKo1 | FuriousGeorge: the code is commented. Look at the comments. |
| 17:34.10 | mishehu | I've been looking thru the mailing list archives and voip-info pages, and I don't know if there is any update to the polycom presence/buddy limit on the 60x's. Does anybody know if the limit has been removed so a full attendant sidecar solution will function? |
| 17:34.12 | eKo1 | evisu: read up on how to program dialplans |
| 17:34.30 | evisu | yep i have but there is quite a lot to go through |
| 17:34.58 | FuriousGeorge | eKo1: you make it sound so easy, i'll look at the code and tell you what i think, and you tell me if im right or wrong |
| 17:35.07 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 17:35.14 | eKo1 | FuriousGeorge: hehe, it won't be easy |
| 17:35.53 | *** join/#asterisk svenna_ (n=svenna@p548D1E25.dip0.t-ipconnect.de) |
| 17:36.05 | FuriousGeorge | which goes back to my original point: ive never found a good source online for further explanaition of some of these error codes |
| 17:36.55 | evisu | no chance any of you can correct my 1 line of code ? :P |
| 17:37.19 | eKo1 | evisu: sure, but it is better that you discover the correction yourself. |
| 17:39.06 | FuriousGeorge | anyone using sangoma cards? |
| 17:39.26 | joburg | Sangoma A101 |
| 17:39.42 | eKo1 | I have a two * boxen, A and B. I have a sip extension X with canreinvite=no. When X makes a call, it goes through A which then goes to B which is connected to my PRI. Why do I get one-way audio though? |
| 17:39.51 | joburg | evisu : where did your code come from ? |
| 17:39.54 | FuriousGeorge | joburg: thats the analog card right? can you comment on my reliability? |
| 17:40.20 | evisu | was trying to put it together myself, .. the second line was already there... .workes fine |
| 17:40.25 | eKo1 | oops, I messed up. I get one-way audio with canreinvite=yes. |
| 17:40.27 | joburg | FG : no the E1 card ( EUROISDN ) |
| 17:40.28 | evisu | i just added the wait() |
| 17:40.35 | eKo1 | The problem disappears with canreinvite=no. |
| 17:40.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:40.54 | *** join/#asterisk razu_ (n=razu@87-119-182-130.tll.elisa.ee) |
| 17:41.52 | joburg | evisu : try : exten => _011.,1,Wait(10) exten => _011.,2,Macro(dialout-trunk,3,${EXTEN},, |
| 17:42.00 | [TK]D-Fender | FuriousGeorge: No, the A101 is the 1 port T1/E1. The A200 is their anaolg card and it works great. |
| 17:42.09 | evisu | many, many thanks joburg |
| 17:42.15 | [TK]D-Fender | joburg: Getting COLDER... |
| 17:42.20 | [TK]D-Fender | evisu: That won't do it... |
| 17:42.36 | joburg | i concur with Sangoma working great |
| 17:42.38 | [TK]D-Fender | evisu: You want it to wait for DIALTONE for a few seconds before sending the DTMF..... |
| 17:42.44 | FuriousGeorge | [TK]D-Fender: did you use a tdm400p before then or at any point, especially for fxo? |
| 17:42.46 | *** join/#asterisk vader192 (n=vader192@65.174.123.126) |
| 17:43.03 | [TK]D-Fender | FuriousGeorge: Yup, owned one myself and consulted for customers possessing both. |
| 17:43.03 | evisu | yeah i see what you're sayin TKD |
| 17:43.21 | [TK]D-Fender | evisu: Dial(Zap/g1/wwwww12345) |
| 17:43.28 | vader192 | Can anyone answer a question about a 4 span t1 card for me? |
| 17:43.44 | [TK]D-Fender | evisu: add "w"'s in front of your number. I don't know what to delay is for cetian, but I think it was .5 sec each |
| 17:43.59 | [TK]D-Fender | certain* |
| 17:44.01 | FuriousGeorge | [TK]D-Fender: my hair has started falling out and im pretty sure my tdm400 is the cause, is that possible? |
| 17:44.32 | evisu | will give that one a try... really many thanks TKD |
| 17:44.34 | [TK]D-Fender | FuriousGeorge: No, the FCC has cleared Digium of radioactive emmissions charges that have been previously levied against them.... |
| 17:44.52 | file | radioactive emissions are bad, mmmk? |
| 17:44.54 | FuriousGeorge | i dont trust the government |
| 17:45.01 | [TK]D-Fender | FuriousGeorge: Maybe you could describe the probelms you're haveing that may contribute to hair loss ;) |
| 17:45.19 | joburg | evisu : exten => _011.,1,Wait(10) then exten => _011.,2,Dial(Zap/g1/${EXTEN}) |
| 17:45.26 | [TK]D-Fender | file: Another card, fresh from the particle accelerator! |
| 17:45.36 | vader192 | Can anyone tell me if its possible to use a single dchan on a 4 span card for PRI? |
| 17:45.39 | joburg | i tried and tested it - it owrks |
| 17:45.42 | [TK]D-Fender | joburg: AGAIN, not what he's looking to do... |
| 17:45.59 | FuriousGeorge | i was just refering to my general problem.. i gotta get to the location after hours to check it out more, but all i know about the current problem is that i have a strange hookstate on one of my pstn dids, so it cant receive calls |
| 17:46.15 | [TK]D-Fender | joburg: Yes it waits before dialing, but it PULSS THE LINE IMMEDIATLY. He wasn't it to pull the line THEN wait. Claer? |
| 17:46.21 | FuriousGeorge | [TK]D-Fender: luckily it works for outbound calling, since its the first in my zap group |
| 17:46.29 | mishehu | [TK]D-Fender: you're somewhat familiar with the polycoms no? if so, do you know if they have yet removed the presence limit of 8 on the soundpoint 60x's ? |
| 17:46.38 | [TK]D-Fender | mishehu: Since 1.6.6 |
| 17:46.39 | FuriousGeorge | and thank god its the last one in my incoming trunk |
| 17:46.58 | FuriousGeorge | s/trunk/"hunt group" |
| 17:47.20 | [TK]D-Fender | FuriousGeorge: A200 is very nice I must say. Not quite as solid at faxing last I checked but I always advise seperate analog lines for that anyways. |
| 17:47.43 | FuriousGeorge | i advise the same |
| 17:48.00 | mishehu | [TK]D-Fender: ah, and 1.6.6+ must be quite recent then, since somebody stated back in April that it was still a problem |
| 17:48.11 | [TK]D-Fender | FuriousGeorge: Did you try to ensure the telce is issuing CDS? |
| 17:48.28 | [TK]D-Fender | mishehu: 1.6.6 was out then IIRC... could check the changelog..... |
| 17:48.36 | mishehu | [TK]D-Fender: have you used the 60x's with the sidecar ? |
| 17:48.38 | FuriousGeorge | you mean telco, right? whats cds? |
| 17:48.46 | FuriousGeorge | caller id? |
| 17:48.52 | [TK]D-Fender | mishehu: Yup, 2x fully loaded with presence.... lights up like Christmas ;) |
| 17:48.58 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:48.59 | joburg | evisu : exten => _011.,1,Dial(Zap/g1/) then exten > _011.,2,Wait(10) then _011.,3,SendDTMF(${EXTEN}) |
| 17:49.02 | [TK]D-Fender | FuriousGeorge: Call Disconnect Supervision. |
| 17:49.22 | *** part/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:49.26 | [TK]D-Fender | joburg: NO, and getting colder still.... |
| 17:49.52 | evisu | heh |
| 17:49.57 | FuriousGeorge | [TK]D-Fender: thanks for the suggestion. im gonna look into that |
| 17:49.58 | Assid | [TK]D-Fender: you got 1.6.7 |
| 17:50.13 | mishehu | [TK]D-Fender: good to hear, gets an idiot customer of mine an alternative to some cheezy Iwatsu phone system. ;-) |
| 17:50.41 | Dr-Linux|work | wow |
| 17:50.51 | Dr-Linux|work | 1.6.7 |
| 17:50.55 | hmmhesays | 0(12276) ERROR: fix_nated_sdp: cannot extract body from msg! |
| 17:50.56 | hmmhesays | bah |
| 17:51.06 | Dr-Linux|work | but i'm still looking for 1.4 :( |
| 17:51.14 | Dr-Linux|work | we are behind |
| 17:51.31 | vader192 | No idea on the dchan question huh? |
| 17:51.59 | mishehu | if I'm not mistaken, 1.6.x sip firmware for the polycom soundpoints requires bootrom 3.1 |
| 17:52.03 | Dr-Linux|work | vader192, what was the question? |
| 17:52.16 | MstlyHrmls | mishehu: not necessarially |
| 17:52.28 | vader192 | Is it possible to use 1 dchan on a 4 span card? |
| 17:52.49 | joburg | evisu : line open the ZAp channel line 2 waits for 10 sec line 3 sends the digits that was dialled - it worth a shot.... |
| 17:53.01 | mishehu | MstlyHrmls: what is the min. bootrom it needs? |
| 17:53.01 | MstlyHrmls | Dr-Linux|work: you're still on 1.3.x? :-( |
| 17:53.11 | vader192 | our switch (dms100) can be set to use 1 dchan for up to 20 t1's in one group |
| 17:53.16 | Qwell | vader192: Look for NFAS. |
| 17:53.26 | Dr-Linux|work | MstlyHrmls, i'm on 1.2.x |
| 17:53.28 | Qwell | ~nfas |
| 17:53.29 | jbot | hmm... nfas is "Non-Facility Associated Signaling" FixMe: saves a D channel on PRI's orsomethingorother |
| 17:53.32 | MstlyHrmls | mishehu: I believe it will run on 2.6, but let me double check |
| 17:53.38 | MstlyHrmls | Dr-Linux|work: ! |
| 17:53.42 | Dr-Linux|work | MstlyHrmls, sorry i was talking about Asterisk version .. |
| 17:53.58 | MstlyHrmls | Dr-Linux|work: ahhhh, sorry, missed that |
| 17:54.03 | joburg | vader : that the ONLY way it works ! there is one d-chan for every T1 |
| 17:54.06 | Dr-Linux|work | MstlyHrmls, heh |
| 17:54.07 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 17:54.20 | Dr-Linux|work | anybody is using cisco 7960 phone? |
| 17:54.27 | MstlyHrmls | Dr-Linux|work: that's what I get for only skimming scrollback ;-) |
| 17:54.36 | joburg | vader : and NO if there is 4 spans there will be 4 d-cahnnels |
| 17:54.55 | Dr-Linux|work | my question is, how many calls i can make at same time? it's doesn't allow me more then 3 at same time .. |
| 17:54.59 | eKo1 | joburg: not necessarily |
| 17:55.08 | vader192 | cool |
| 17:55.11 | evisu | hmm, does the wwww work if its a SIP trunk, not zap... ? |
| 17:55.12 | vader192 | thank you! |
| 17:55.19 | mishehu | wow, 1 14 button expansion module costs almost as much as the phone itself |
| 17:55.25 | Qwell | jbot: nfas is "Non-Facility Associated Signaling", a way to use fewer D channels than you have PRIs. ie; 1 D channel for 4 PRIs, instead of the usual 4 |
| 17:55.27 | jbot | ...but nfas is already something else... |
| 17:55.32 | Qwell | jbot: no, nfas is "Non-Facility Associated Signaling", a way to use fewer D channels than you have PRIs. ie; 1 D channel for 4 PRIs, instead of the usual 4 |
| 17:55.37 | jbot | okay, Qwell |
| 17:56.21 | joburg | nfas : thats new - is it available for E1 ? |
| 17:56.29 | Dr-Linux|work | anybody have cisco 7960? |
| 17:56.39 | MstlyHrmls | mishehu: ok, the 430, 601 & 4000 require 3.1.x BootROMs; 30x, 50x, and 600 require 2.6.1 |
| 17:56.56 | postel | Dr-Linux|work: nope, cisco only sold to you |
| 17:57.17 | joburg | i have a cisco 7950 |
| 17:57.23 | Qwell | Dr-Linux|work: chan_skinny ;) |
| 17:57.39 | joburg | not sure if that helps.... |
| 17:57.46 | Dr-Linux|work | Qwell, :) that doesn't work for cisco 7935 |
| 17:57.52 | Qwell | Dr-Linux|work: yet |
| 17:58.08 | Qwell | If I had one, I could fix it up |
| 17:58.12 | joburg | i have to add it's a SIP 7950 |
| 17:58.19 | Qwell | 7950? No such thing |
| 17:58.27 | Dr-Linux|work | Qwell, again, i posted my problem at chan_sccp mailing list, but no answer so for |
| 17:58.37 | Qwell | Dr-Linux|work: yeah, they aren't going to fix it |
| 17:58.40 | Dr-Linux|work | joburg, what's 7950? :S |
| 17:58.43 | Qwell | Sergio is "busy" with other things |
| 17:58.47 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 17:58.50 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 17:59.06 | Dr-Linux|work | Qwell, but he should fix this bug |
| 17:59.14 | Qwell | Dr-Linux|work: I'll say |
| 17:59.32 | mishehu | MstlyHrmls: ah ok, thanks for the info |
| 17:59.38 | mishehu | we'd be getting 601's. |
| 17:59.47 | joburg | it's a cisco 7950 - a cisco SIP phone |
| 17:59.52 | Qwell | joburg: no such thing :) |
| 18:00.04 | Qwell | 7905 maybe? |
| 18:00.09 | Dr-Linux|work | my question, is that my cisco 7960 can do only 3 calls at the same time, so can i increase it or not possible, |
| 18:00.12 | Qwell | (though, I didn't think the 7905 had sip firmware) |
| 18:00.23 | Dr-Linux|work | was a simple question is some one knows the answer |
| 18:01.26 | Dr-Linux|work | joburg, we have a lot of cisco phones, but never heard about 7950 |
| 18:02.26 | *** join/#asterisk luchshiy (n=anonymou@d212-53-104-193.cust.tele2.ch) |
| 18:02.35 | evisu | would Dial(Zap/g1/wwwww12345) doesnt seem to have the wait effect when used on an IAX trunk.. is there another way ? |
| 18:03.48 | evisu | *no would |
| 18:03.53 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 18:03.53 | *** mode/#asterisk [+o russellb] by ChanServ |
| 18:04.19 | joburg | evisu Dialing on a IXA trunk won't be Dial(Zap/g1/wwwww12345) but Dial(IAX2/something/wwwww12345) |
| 18:04.52 | evisu | yep, got that far :) ... it just doesnt seem to wait, so i figure it may only have that effect on zap |
| 18:05.04 | Dr-Linux|work | can we raise the number of concurrent calls that are active on cisco 7960? |
| 18:06.59 | joburg | evisu : i tested it with SendDTMF and it works .... |
| 18:07.26 | evisu | will give it a go now |
| 18:08.36 | *** join/#asterisk bmg505 (n=leon@dsl-146-59-106.telkomadsl.co.za) |
| 18:09.22 | *** join/#asterisk asterisk_bounty (n=pcm@68.159.139.234) [NETSPLIT VICTIM] |
| 18:09.52 | *** part/#asterisk joburg (n=voipmagi@vc-196-207-38-156.3g.vodacom.co.za) |
| 18:11.32 | anonymouz666 | russellb |
| 18:11.34 | clyrrad | for some reason my AGI script wont print to STDERR ie the CML when i use fwrite and fflush - but the AGI does execute anyone know why? |
| 18:11.36 | anonymouz666 | where you been? |
| 18:11.48 | [TK]D-Fender | evisu: You can't delay (nor should there be a reason to) a SIP/IAX call. there IS no tone or anything |
| 18:11.56 | [TK]D-Fender | Assid: I have all SIP/BR releases |
| 18:12.04 | russellb | anonymouz666: moving |
| 18:12.14 | [TK]D-Fender | Assid: 2.0 is due momentarily |
| 18:13.11 | *** join/#asterisk oej (n=oej@63.117.53.60) |
| 18:13.42 | hmmhesays | so nathelpers fix natted registration doesn't work worth a shiat |
| 18:14.25 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 18:14.28 | evisu | TKD, I know, its strange, but I do need it for my application |
| 18:14.47 | TheCops | Hi, someone can confirm me that TDM2400P are compatible with IBM xseries 226? |
| 18:15.24 | Qwell | TheCops: digium.com has a hardware incompatibility page. |
| 18:15.32 | [TK]D-Fender | evisu: Use a .call file and then use your dialplan to enact the wait & send, then forward the call to the remote (calling) end. |
| 18:15.34 | Qwell | generally, if it's PCI 2.2, it'll work |
| 18:15.42 | TheCops | ok |
| 18:15.46 | TheCops | Qwell, it is 2.2 |
| 18:15.47 | Qwell | but there are some specific servers/motherboards that just...don't |
| 18:15.52 | [TK]D-Fender | evisu: It'll become a "call-back mechanism. |
| 18:15.54 | Qwell | like a couple Dells, I believe |
| 18:16.12 | TheCops | Qwell, yeah I see IBM have a couple too. IBM x345. That's why I want to use the 226 |
| 18:16.18 | [TK]D-Fender | evisu: Unless the "D" option actually does work.... but again, no idea how to delay it. |
| 18:16.58 | TheCops | Qwell, look at this: http://uk.shopping.com/xPF-IBM_IBM_XSERIES_226_Express_x226_Xeon_3_0GHz_800MHz_2MB_L2_512MB_U320_HS_O_Bay, This is what I want to buy, I see PCI 2.2 compliant, but not sure about PCI card type |
| 18:17.04 | TheCops | PCIX and stuff like that |
| 18:17.18 | evisu | aha.. ok I'll look along these lines .. cheers |
| 18:17.18 | TheCops | I'm not really an hardware guys, difficulties to determine what kind of pci I need |
| 18:17.24 | Qwell | TheCops: I'm not going to say it works or not. :) |
| 18:17.32 | Qwell | because I simply do not know |
| 18:17.56 | SplasPood | So whats the deal with toll free numbers and answer supervision... Do a lot of them not ANSWER until yer out of the IVR to try and get out of paying for that call time? |
| 18:18.10 | Qwell | SplasPood: some, yeah |
| 18:18.12 | hmmhesays | anyone familiar with nathelper in ser? |
| 18:18.14 | Dr-Linux|work | TheCops, make sure you are buying the correct cards according to your slots .. |
| 18:18.18 | Qwell | I think somebody specifically mentioned...what was it...amex? |
| 18:18.42 | Dr-Linux|work | TheCops, look there, your slots are 3.3v or 5v |
| 18:18.45 | TheCops | Qwell, yeah I know, but can you confirm me that "technical" specs SHOULD work? |
| 18:19.10 | anonymouz666 | Qwell: what kind of servers (rack) do you know that works 100% with digium hardware? |
| 18:19.15 | TheCops | Dr-Linux|work, dont know |
| 18:19.15 | Qwell | TheCops: as long as it has full height(/length?) PCI 2.2, and molex connectors... |
| 18:19.20 | Qwell | it *should* work, but... |
| 18:19.23 | TheCops | ok |
| 18:19.25 | TheCops | yeah I know |
| 18:19.26 | anonymouz666 | dell poweredge has some problems with e1000 |
| 18:19.57 | TheCops | Qwell, molex connectors? |
| 18:20.04 | Qwell | TheCops: PC power cable standard |
| 18:20.07 | TheCops | ok |
| 18:20.09 | TheCops | duh |
| 18:20.10 | Qwell | the 4 pin dealies |
| 18:20.13 | TheCops | yeah yeah |
| 18:20.18 | TheCops | thanks :P |
| 18:20.22 | Qwell | those are needed for FXS |
| 18:20.28 | Dr-Linux|work | anonymouz666, we have 4 production asterisk servers and 2 test server, all those are Dell poweredge |
| 18:20.42 | anonymouz666 | 850? |
| 18:20.57 | Dr-Linux|work | hhm.. |
| 18:21.06 | Dr-Linux|work | i never use molex connection .. |
| 18:21.07 | anonymouz666 | do you have onboard gigabit card enabled? |
| 18:21.16 | E-bola | u dont use molex... |
| 18:21.17 | E-bola | lol |
| 18:21.28 | yatesy | heh |
| 18:21.31 | *** join/#asterisk `Kevin (n=Kevin@64.243.236.20) |
| 18:21.33 | clyrrad | for some reason my AGI script wont print to STDERR ie the CML when i use fwrite and fflush - but the AGI does execute anyone know why? |
| 18:21.41 | Dr-Linux|work | if card is getting supply from bus, why i need molex connector |
| 18:21.45 | clyrrad | Its like its lots its connection to print on the CLI.... |
| 18:22.12 | yatesy | because it needs more power i'm guessing Dr-Linux|work |
| 18:22.25 | Assid | [TK]D-Fender: should i just wait for 2.0? |
| 18:22.36 | [TK]D-Fender | Assid: What are you missing right now? |
| 18:22.38 | clyrrad | I define it like this define('STDERR',fopen('php://stderr', 'w')); |
| 18:22.56 | *** join/#asterisk soylentgreen (n=fgast@nebukadnezar-em0.only640k.org) |
| 18:22.59 | Assid | nothing as such.. but who knows , some bug fixes etc? |
| 18:23.01 | FuriousGeorge | [TK]D-Fender: interestingly i saw some stuff on line about sbc not supporting cds in certain areas, so i looked i tried contacting att, which is somehow merged with sbc, but i had to file a trouble ticket cuz no one knew. so a tech will get back to me re: CDS |
| 18:23.27 | Dr-Linux|work | anonymouz666, i have molex connector jack in my digium cards, also digium cards have jack for molex connector, but i never use that .. everything works fine for me so far |
| 18:23.35 | clyrrad | ...anybody...? |
| 18:23.42 | Dr-Linux|work | yatesy, more power? |
| 18:23.49 | [TK]D-Fender | FuriousGeorge: Telcos = stupid much of the time. Poeple answering the phone don't know what services they can even offer |
| 18:23.51 | Qwell | E-bola: Yes you do, on FXS cards |
| 18:24.00 | Qwell | That's what provides ring voltage to the phones |
| 18:24.17 | yatesy | thought so |
| 18:24.18 | E-bola | Qwell: huh? |
| 18:24.23 | Qwell | E-bola: molex |
| 18:24.28 | E-bola | qwell: hehe yes? |
| 18:24.38 | E-bola | i think ur adressing the wrong person :) |
| 18:24.43 | Qwell | <E-bola> u dont use molex... |
| 18:24.49 | FuriousGeorge | [TK]D-Fender: well you may have been onto something, ill let you know what they say |
| 18:24.54 | E-bola | it was ironic hence the ... |
| 18:25.17 | E-bola | i thought he meant he never used molex connectors in computers |
| 18:25.23 | E-bola | which is funny cuz the u cant have any drives at all |
| 18:25.29 | E-bola | the=then |
| 18:25.30 | [TK]D-Fender | Qwell: And theta why your shirts are all soo crinkly! ;) |
| 18:25.38 | *** join/#asterisk BZBW (n=wlwzhang@ip67-153-142-109.z142-153-67.customer.algx.net) |
| 18:25.50 | [TK]D-Fender | jkhdfsahjlkdfsalhjkdasfhkklhjfdhjfds |
| 18:25.52 | Dr-Linux|work | E-bola, aww, i appritiate your thinking :) |
| 18:25.54 | E-bola | qwell: american by any chance? :0) |
| 18:25.59 | [TK]D-Fender | CAN'T TYPE TODAY DAMMIT |
| 18:26.03 | Dr-Linux|work | [TK]D-Fender, relax |
| 18:26.20 | FuriousGeorge | the polarity reversal in my CLI refer to inverting where the green and the red line are "supposed to go" relative to the TDM? |
| 18:26.26 | FuriousGeorge | *FXO |
| 18:27.01 | mocker | Crap. I don't think cdr_mysql contains all the info my cdr_custom does. |
| 18:27.02 | mocker | :( |
| 18:27.16 | Dr-Linux|work | [TK]D-Fender, hhm.. don't worry after weekend such things happen around :) |
| 18:27.47 | [TK]D-Fender | FuriousGeorge: Something like that . |
| 18:28.09 | hmmhesays | ~seen file |
| 18:28.12 | jbot | file is currently on #openezx. Has said a total of 70 messages. Is idling for 43m 20s, last said: 'radioactive emissions are bad, mmmk?'. |
| 18:28.26 | file | hola |
| 18:28.42 | FuriousGeorge | [TK]D-Fender: is it something that could cause wierdness? fxo being reported as in use when there should be no active channel there, incoming calls not being bridged correctly for a "Strange hook state" that sort of thing |
| 18:28.50 | hmmhesays | hey file, you always seem to give me some insight on SER questions, I have another |
| 18:28.52 | E-bola | hi file |
| 18:29.15 | [TK]D-Fender | FuriousGeorge: I am thinking that better disconnect detection may help reduce dead channels.... |
| 18:29.16 | hmmhesays | you ever use nathelper module? |
| 18:29.16 | file | ask and you might receive |
| 18:29.19 | SplasPood | Qwell: yea, Amex and american airlines seem to do it, at least |
| 18:29.19 | file | yes |
| 18:29.33 | SplasPood | Qwell: interesting.. seems like a great scam... |
| 18:29.42 | FuriousGeorge | the former i can fix by restarting asterisk and waiting for the dialtone to reset, the latter is still happening after restarting the server iteself |
| 18:29.45 | Qwell | SplasPood: nah, it's technical legal, I'd think |
| 18:30.00 | Qwell | technically too |
| 18:30.02 | hmmhesays | if I use fix_nated_register(); it doesn't seem to rewrite anything when I save the location |
| 18:30.03 | SplasPood | Qwell: Well a scam might be legal :) |
| 18:30.09 | rg1_ | exten => s,n,Set(U_TEMP_USER_RESPONSE=${IF($[${TEMP_SPEECH_SCORE} < ${TQM_SPEECH_MIN_ACCEPT_SCORE}]?${TQMUR_BAD_RESPONSE}:${TEMP_USER_RESPONSE})}) |
| 18:30.15 | SplasPood | Qwell: Doesn't make it less of a scam :) Legal scams are the best kind |
| 18:30.17 | hmmhesays | i look at the entry in usrloc and it still has the private IP |
| 18:30.23 | Qwell | rg1_: That's hardcore |
| 18:30.26 | rg1_ | Can someone tell me if they see anything wrong with the syntax of that? |
| 18:30.57 | TheCops | Qwell, 33 100 133mhz depend on something with TDM2400P Card? |
| 18:31.01 | SplasPood | Qwell: Its funny.. If you google for "toll free" and "answer supervision" you get a ton of different telco's TOS statements saying the customer MUST provide it |
| 18:31.16 | hmmhesays | unless i'm missing the point of fix_nated_register() |
| 18:31.34 | Qwell | rg1_: looks okay, first pass |
| 18:31.42 | file | hmmhesays: it should rewrite the contact... but I haven't done SER stuff in a long time |
| 18:31.44 | rg1_ | exten => s,n,Set(U_TEMP_USER_RESPONSE=${IF($[400 < 300]?100:200)}) |
| 18:31.49 | rg1_ | theres a simpler version |
| 18:31.57 | Qwell | looks just fine |
| 18:32.04 | rg1_ | U_TEMP_USER_RESPONSE is getting set to 100 - I would think it should be 200 |
| 18:32.17 | Qwell | rg1_: yes, it should be 200 |
| 18:32.34 | hmmhesays | it sure seems to not re-write the contact |
| 18:33.24 | Qwell | rg1_: try to noop the expression, instead of setting the var |
| 18:34.23 | anonymouz666 | hmmhesays: fix_nated_contact()? |
| 18:36.36 | *** join/#asterisk ghotiboy1 (n=ghotiboy@24-176-46-6.dhcp.klmz.mi.charter.com) |
| 18:37.00 | ghotiboy1 | hi there...does anyone know if I can run asterisk in VMWare and use Digium cards? |
| 18:37.14 | ghotiboy1 | i've seen conflicting info on google searches |
| 18:37.19 | Qwell | ghotiboy1: I wouldn't do it, personally |
| 18:37.31 | ghotiboy1 | any reasons? timing issues? |
| 18:37.35 | Qwell | asterisk is a realtime application |
| 18:37.36 | Qwell | exactly |
| 18:38.01 | ghotiboy1 | if i were to set the vmware session as real-time would that help? |
| 18:38.07 | ghotiboy1 | is it even possible? |
| 18:38.08 | clyrrad | I am having trouble getting my AGI to print to the CLI - I have define('STDERR',fopen('php://stderr', 'w')); but when I try to print to STDERR nothing gets printed on the CLI - but if I rund the agi script from the command line I get the output from the print to STDERR - how can I interface STDERR with the asterisk CLI for debugging output on AGI scripts? |
| 18:38.09 | Qwell | not really |
| 18:38.35 | ghotiboy1 | ok, well, that pretty much settles it |
| 18:38.36 | ghotiboy1 | thanks |
| 18:38.47 | Qwell | ghotiboy1: I mean, don't get me wrong |
| 18:38.54 | Qwell | people have had success with vmware and/or xen, but... |
| 18:39.10 | Qwell | If you're going to do any volume of calls, I would highly recommend using a dedicated machine |
| 18:39.30 | intralanman | clyrrad: i know i saw something on voip-info about that |
| 18:39.38 | ghotiboy1 | it would be dedicated...just virtualized to make for easier backup/recovery/redeployment |
| 18:39.40 | clyrrad | do you have a link? |
| 18:39.44 | clyrrad | its been driving me nuts |
| 18:39.46 | intralanman | nope |
| 18:39.47 | intralanman | google |
| 18:39.49 | Qwell | ghotiboy1: there are better ways of doing backups and such :) |
| 18:39.50 | clyrrad | it was working before |
| 18:40.03 | clyrrad | printing to the CLI - but for some reason today it does not |
| 18:40.15 | ghotiboy1 | Qwell: depends on clients current environment |
| 18:40.17 | *** join/#asterisk UlbabraB (n=UlbabraB@host241-43-static.72-81-b.business.telecomitalia.it) |
| 18:40.24 | intralanman | clyrrad: was asterisk restarted recently? |
| 18:40.25 | Qwell | ghotiboy1: yes, of course |
| 18:40.32 | ghotiboy1 | we want to integrate, not redesign |
| 18:40.35 | clyrrad | yes |
| 18:40.35 | BZBW | hi, I have 2 FXO GW, each will dial in to a virtual extension, say 444 and 666, and each of these extensions will then prompt user for different action, but it seems all calls go into extension 666, check many times with no clue, any idea? |
| 18:40.59 | clyrrad | intralanman - yes iti was restarted... |
| 18:41.25 | intralanman | ahh.... probably need to change the switches it was started with |
| 18:41.31 | intralanman | as i recall |
| 18:41.40 | ghotiboy1 | another question...different topic...I have some polycom phones and want to accept multiple calls to the same extension...this doesn't seem to happen...just gets forwarded to VM...is there a setting for each extension? |
| 18:41.42 | clyrrad | i have it started with vvvvvvvvvvgc |
| 18:42.22 | ghotiboy1 | the phones are 3-line deals |
| 18:42.47 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 18:42.50 | clyrrad | intralanman - which switch am i missing? |
| 18:43.00 | intralanman | not sure.... i don't remember that well |
| 18:43.08 | clyrrad | does anyone else know? |
| 18:43.56 | *** join/#asterisk pfn (n=pfnguyen@netblock-66-245-252-239.dslextreme.com) |
| 18:44.15 | clyrrad | hrm... this is frustrating... |
| 18:44.23 | intralanman | clyrrad: http://www.voip-info.org/wiki-Asterisk+AGI |
| 18:45.03 | ghotiboy1 | anyone know if you can (by default) accept multiple calls to the same extension? if not, how do I config that for the extensions? |
| 18:45.08 | intralanman | clyrrad: if that doesn't help, then i don't know |
| 18:45.52 | clyrrad | yea it dont answer my question |
| 18:45.54 | clyrrad | thanks anyway |
| 18:46.38 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 18:47.29 | [TK]D-Fender | ghotiboy1: Yes you can receive MANY calls against a single registration on all Polycom models. |
| 18:48.06 | ghotiboy1 | hmmm....so for some reason it isn't forwarding the calls from asterisk... |
| 18:48.20 | [TK]D-Fender | ghotiboy1: : Its all in how you set up the phone. |
| 18:48.33 | ghotiboy1 | so it IS the phone setup then |
| 18:48.37 | ghotiboy1 | I thought it was |
| 18:48.59 | [TK]D-Fender | ghotiboy1: And don't think of it as the phone forwarding anything. Its * gigving up and then send the call on to "wherever". |
| 18:49.31 | ghotiboy1 | yeah...i figured that the phone was either not responding or sending back a rejection |
| 18:49.55 | [TK]D-Fender | ghotiboy1: I have some of mine using 1 line key per registration (and using all 3 on my IP 501) and allowing each of those to support 5 calls each. That means I can juggle up to 1`5 calls at a time on that setups |
| 18:50.15 | [TK]D-Fender | ghotiboy1: Yes, that would be a better description. |
| 18:50.20 | ghotiboy1 | ah...ok |
| 18:50.27 | ghotiboy1 | perfect...thanks |
| 18:50.28 | [TK]D-Fender | ghotiboy1: At which point * continues on to do whatever you told it to. |
| 18:50.30 | *** join/#asterisk rbordeaux (i=hidden-u@80.169.196.234) |
| 18:50.53 | [TK]D-Fender | ghotiboy1: You want to focus on "CallsPerLineKey" and "NumLineKeys" in your Polycom config. |
| 18:51.14 | [TK]D-Fender | ghotiboy1: How many do you won, and what models? |
| 18:51.16 | [TK]D-Fender | own* |
| 18:51.31 | *** join/#asterisk romdav (n=romdav@201.155.183.225) |
| 18:51.53 | romdav | Hi every one |
| 18:51.56 | *** join/#asterisk Un1x (n=x@CPE001731208485-CM0011ae8a7b0a.cpe.net.cable.rogers.com) |
| 18:51.59 | *** join/#asterisk DaKalle (i=kvirc@p54995D23.dip.t-dialin.net) |
| 18:52.35 | romdav | i need help on make work a g729 licnces |
| 18:52.57 | denon | you get free support from digium on installing them |
| 18:53.08 | hmmhesays | fix_nated_register does not save the registration in the usrloc database |
| 18:53.09 | ghotiboy1 | [TI]D-Fender: not sure...a friend is setting it up and I'm consulting...I think 20 phones...some 430 and some 501 (I think) |
| 18:54.13 | *** part/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:54.20 | *** join/#asterisk mog (i=ejabberd@c-71-207-215-93.hsd1.al.comcast.net) |
| 18:54.20 | *** mode/#asterisk [+o mog] by ChanServ |
| 18:54.25 | *** join/#asterisk ki2k (n=ki2k_@207.231.83.242) |
| 18:54.33 | ki2k | wow, big channel today |
| 18:55.10 | ki2k | anyone up? |
| 18:55.42 | file | perhaps |
| 18:55.47 | ki2k | heh |
| 18:56.32 | ki2k | are there any 'required' contexts in extensions.conf besides what you define in your sip.conf or iax.conf or zap.conf ? |
| 18:56.49 | ki2k | is general and globals required? |
| 18:57.05 | *** join/#asterisk zedkatuf (n=zedkatuf@82-32-57-69.cable.ubr08.azte.blueyonder.co.uk) |
| 18:58.05 | zedkatuf | hi all, what do I need to add to my sp.conf so that when someone phones me, they hear a ring tone (ie a "drrng", or in my case as I'm in the UK, a "drrng" "drrrng") ? |
| 18:58.11 | zedkatuf | sp/sip |
| 18:58.41 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 18:58.44 | *** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no) |
| 18:58.47 | romdav | I have a g729 licences and work ok on sip to sip, but not work on sip to asterisk i can't hear the asterisk messages on mailbox check, how i can fix it? |
| 19:00.10 | *** join/#asterisk tlow (i=tlowe@gateway/tor/x-9ada5524ae7c9e9f) |
| 19:00.18 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 19:00.51 | *** part/#asterisk EmleyMoor (i=ejabberd@hallam.tinsleyviaduct.com) |
| 19:01.20 | *** join/#asterisk HeMan (n=jimmy@1-1-7-40a.far.sth.bostream.se) |
| 19:02.44 | DaKalle | i think i have got a problem with the dial-command on my asterisk machine: weh someone from outside is calling my asterisk machine, asterisk lets my internal sip phones ring, but they aren't displaying the caller id, but the internal extension-number of themselves |
| 19:03.02 | DaKalle | weh/when |
| 19:03.56 | HeMan | Hi! I have a SIP account (digisip) that i can connect with and i can recieve calls and forward them to my softphone (ekige) but i have no sound |
| 19:04.07 | HeMan | any clues what that could be? |
| 19:04.15 | *** join/#asterisk alexhopper (n=a27386@CPE000103d29ae2-CM001225dfdfe0.cpe.net.cable.rogers.com) |
| 19:04.46 | DaKalle | portforward ? |
| 19:06.54 | DaKalle | i had the same problem and i "fidex" it in forwarding all ports to my asterisk-machine |
| 19:07.05 | [hC] | Interesting. Clients complaining of their polycom ip501's being too quiet when they're on a call, via voip, over my pri... Ive never had audio level complaints before, I wonder what else could be contributing.. |
| 19:07.09 | HeMan | DaKalle: i have a NAT firewall but i run linux 2.6.18rc4 with sip-helper |
| 19:07.44 | DaKalle | ok |
| 19:07.56 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 19:08.05 | [TK]D-Fender | [hC]: You can up the gains direct in their phone configs and/or set the volumes to "sticky" at the very least. |
| 19:08.59 | [hC] | [TK]D-Fender: I've set them to sticky already, but left the gains at default. Do you usually edit anything in sip.cfg aside from the main proxy address, sticky, MWI stuff? |
| 19:09.17 | [hC] | I followed the a@h suggestions for sip.cfg, I never modify anything else really. |
| 19:09.37 | [hC] | NTP, SIP Server, MWI stuff, sticky volume, and I think something to do with Intercom, im not sure. |
| 19:10.14 | *** join/#asterisk murf (n=steve_mu@216.166.159.235) |
| 19:10.46 | *** part/#asterisk Skaag (n=hintza@212.199.180.157.static.012.net.il) |
| 19:11.53 | hmmhesays | what the hell happened to iptel.org? |
| 19:12.27 | anonymouz666 | Use OpenSER |
| 19:12.33 | DaKalle | has anyone an answer to my dial-command problem ? |
| 19:14.51 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 19:16.05 | HeMan | hmm, seems that the sip-helper wont do the, umm, rtp-ports |
| 19:16.25 | *** join/#asterisk ki2k (n=ki2k_@207.231.83.242) |
| 19:16.28 | ki2k | hi |
| 19:16.57 | ki2k | my xlite doesnt know the other side has hung up. Is that an asterisk or xlite issue? |
| 19:17.30 | zedkatuf | hi all, what's the technical term for the "drrng" noise that someone hears in their phone when they dial a number? |
| 19:17.42 | hmmhesays | anonymouz666: a lot of ser's base documentation applies to openser and is helpful |
| 19:17.59 | file | zedkatuf: ringing tone? |
| 19:18.00 | alexhopper | ring tone? |
| 19:19.00 | zedkatuf | ok, thanks.....when I try my sip phone, I don't hear a ringing tone at all......just silence.....am trying to google the answer atm |
| 19:19.25 | [TK]D-Fender | [hC]: There is a bunch of little things I do in there, and I usually leave ALL of the reg stuff in the phone file, not sip.cfg main. |
| 19:19.26 | hypa7ia | [hC]: yo |
| 19:19.43 | [TK]D-Fender | [hC]: But I don't typically mess with initial gains. |
| 19:20.02 | [TK]D-Fender | ki2k: * |
| 19:20.18 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 19:20.58 | ki2k | [TK]D-Fender: what do you think it may be? |
| 19:21.01 | file | zedkatuf: what's the console output like? |
| 19:21.11 | kpettit | can anybody recommend any other good SIP/IAX providers other than Teliax? |
| 19:21.25 | *** join/#asterisk mfarley (i=mfarley@puffer.wheatstate.net) |
| 19:22.57 | [hC] | [TK]D-Fender: Do you know off the top of your head what else you do? Could prove useful! :) |
| 19:23.01 | [hC] | hypa7ia: heyy, how goes |
| 19:23.24 | hypa7ia | [hC]: it goes great |
| 19:23.33 | hypa7ia | trying to solve a polycom line appearance mystery |
| 19:23.51 | [TK]D-Fender | [hC]: Key remapping, global MicroBrowser settings, custom ring-tone's, and so much more. |
| 19:23.56 | hypa7ia | the user doesn't know what he did, but now there are animated arrows by his speeddials and they aren't in the manual :s |
| 19:24.28 | hypa7ia | haha |
| 19:24.31 | Assid | i still gotta work on key remapping |
| 19:24.34 | Assid | hehe |
| 19:24.53 | zedkatuf | file: shall I pastebin output? |
| 19:24.54 | hypa7ia | <PROTECTED> |
| 19:24.59 | file | zedkatuf: yes |
| 19:25.08 | Assid | need to remap "hold" on 301 for contact list |
| 19:25.29 | [TK]D-Fender | [hC]: Also do things like disabling the HTTP admin page so morons don't fsck up my configs ;) |
| 19:25.36 | [hC] | [TK]D-Fender: Gotcha. :) |
| 19:25.46 | *** join/#asterisk rpm (n=russell@S01060002b3d10d24.cg.shawcable.net) |
| 19:25.51 | Assid | oh yeah.. gotta enable that.. :P |
| 19:25.59 | TheCops | Someone can recommend me a good model (around 2000-4000$) of a Dell server for using TDM2400P |
| 19:26.20 | intralanman | why dell? yuk |
| 19:26.21 | [TK]D-Fender | TheCops: Big card.... hrm... |
| 19:26.35 | TheCops | [TK]D-Fender yup big card hehe |
| 19:26.39 | TheCops | Fully-length card |
| 19:27.14 | rpm | Has anyone had any success with the Authenticate() application and jumping to priority n+101 on failure? It seems to never jump to the next priority but hang up my current call. I have put some debugging messages in the app_authenticate.c source to log some warnings. It is jumping to priority inward-dial|s|104 which I have set to jump to my default contexts IVR system. |
| 19:27.16 | *** join/#asterisk }btorch{ (n=btorch@208.63.19.179) |
| 19:28.16 | TheCops | [TK]D-Fender, I have some difficulties to found a good server that I willn't have bug or stuff like that |
| 19:31.51 | *** join/#asterisk num000 (n=numerobi@e177180054.adsl.alicedsl.de) |
| 19:32.16 | Juggie | Hello Class! |
| 19:32.32 | hypa7ia | Juggie!!! |
| 19:32.42 | Juggie | oooo look who has re-apeared! |
| 19:33.09 | hypa7ia | lol |
| 19:33.17 | *** join/#asterisk mountainm2k (n=mountain@216.87.64.218) |
| 19:33.28 | hypa7ia | how goes my friend |
| 19:34.08 | Juggie | not too bad, working hard :) |
| 19:34.13 | Juggie | are you going to astricon this year? |
| 19:34.51 | *** join/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 19:36.27 | hypa7ia | maybe... we'll see :) |
| 19:36.58 | Qwell | hypa7ia: hey |
| 19:37.05 | hypa7ia | hi :) |
| 19:37.42 | *** join/#asterisk ApEtc (i=apetc@ip70-162-197-214.ph.ph.cox.net) |
| 19:37.48 | Qwell | hypa7ia: see /msg |
| 19:41.05 | blitzrage | hypa7ia: hey hey |
| 19:41.15 | hypa7ia | sup blitzrage |
| 19:41.17 | rpm | http://rafb.net/paste/results/NLPJ1X49.html, is the problem I am having with the Authenticate function. |
| 19:41.32 | blitzrage | typo's suck |
| 19:41.57 | obiwanmikenolte | Has anyone had a problem connecting to Asterisk 1.2.10 while it's running? It's not creating /var/run/asterisk.ctl or /var/run/asterisk.pid for some reason. I tried running asterisk as root, and it still doesn't work, so it's not a permissions problem (root can write to /var/run with no problem). I've installed earlier versions without this problem, so it may be a bug, but has anyone else experienced it? |
| 19:41.59 | ki2k | is there such a thing as SIPBarge? like ZapBarge? |
| 19:42.35 | obiwanmikenolte | ChanSpy? |
| 19:42.56 | obiwanmikenolte | Sorry. ki2k: chanspy? |
| 19:43.02 | ki2k | ok looking |
| 19:43.05 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 19:43.06 | *** mode/#asterisk [+o russellb] by ChanServ |
| 19:43.23 | obiwanmikenolte | But it's one-way audio |
| 19:43.31 | obiwanmikenolte | You want to break into a conversation? |
| 19:43.33 | ki2k | meaning they cant hear you? |
| 19:43.34 | ki2k | no |
| 19:43.38 | ki2k | well, both would be nice |
| 19:43.43 | obiwanmikenolte | They can't hear you |
| 19:43.46 | ki2k | ok |
| 19:43.56 | obiwanmikenolte | Yeah, I know. I don't know of an application that lets you break in and talk |
| 19:43.56 | ki2k | what would be nice is the ability to wisper into the user's ear too |
| 19:44.04 | ki2k | the far end cant hear |
| 19:44.06 | obiwanmikenolte | You running a 900 operation? |
| 19:44.10 | ki2k | no |
| 19:44.12 | ki2k | hahah |
| 19:44.14 | obiwanmikenolte | : ) |
| 19:44.26 | obiwanmikenolte | Yeah, that'd be sweet. |
| 19:44.27 | ki2k | call customer service training |
| 19:44.28 | *** join/#asterisk jcwunder (n=chris@ppp-62-245-160-41.dynamic.mnet-online.de) |
| 19:44.50 | ki2k | so a supervisor can wisper into the agent's ear to advise him of what to do |
| 19:45.00 | ki2k | the customer cant hear |
| 19:45.13 | russellb | ChanSpy in 1.4 has a channel whisper mode |
| 19:45.14 | russellb | :) |
| 19:45.20 | ki2k | twhoa |
| 19:45.26 | anonymouz666 | yeap |
| 19:45.27 | anonymouz666 | I saw that |
| 19:45.46 | ki2k | where do you see this info? |
| 19:45.51 | ki2k | in the svn? |
| 19:45.53 | anonymouz666 | russellb kevin wrote it? |
| 19:46.01 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 19:46.02 | russellb | yeah |
| 19:46.50 | *** join/#asterisk harlequin516 (n=sham@dsl01-ppp-4444.fastq.com) |
| 19:46.54 | harlequin516 | What is Auto fallthrough ? |
| 19:47.22 | intralanman | it's what happens when you walk on thin ice :) |
| 19:47.43 | intralanman | no configuration needed |
| 19:48.41 | *** join/#asterisk klork (i=vny@h678631.serverkompetenz.net) |
| 19:48.59 | *** join/#asterisk Katty (n=aisaacs@64.82.232.54) |
| 19:49.06 | Katty | hihi |
| 19:49.39 | klork | hullo guys m setting up asterisk server for my home ..... and hav linux and windows machines........ and wanna know which clients are known to best work with asterisk |
| 19:50.03 | klork | any links / ideas ? |
| 19:50.17 | ki2k | when is asterisk 1.4.x gonna get released? |
| 19:50.24 | obiwanmikenolte | Good question. |
| 19:50.28 | ki2k | heh |
| 19:50.39 | Katty | when you least expect it. |
| 19:50.40 | *** join/#asterisk vosque (i=bj3jhuqz@vac.vis.nu) |
| 19:51.30 | generalhan | hwy guys, ive got an easy one for you today. im trying to get my voicemail boxes setup so a caller can hit 0 during the message to get to an operator, but i can figure out where to define the "o" extension, in which context i mean |
| 19:52.04 | vosque | can asterisk be set up to listen to multiple sip ports? |
| 19:52.23 | obiwanmikenolte | ki2k: I'm not sure if this is what you're asking, but if you're in the cli and type "show application <name>", you can look at the syntax and function of applications. The Good Book has a lot of them, but like 50 have been added since it was published |
| 19:52.35 | mut | got a question, i got this 1gig stick of ram, its detected as 512mb with one motherboard i have and 1gig with a dell pc |
| 19:52.39 | mut | anyone know why itd do that |
| 19:52.49 | *** part/#asterisk HeMan (n=jimmy@1-1-7-40a.far.sth.bostream.se) |
| 19:52.50 | Juggie | generalhan, in the same context as where the voicemail starts |
| 19:52.55 | Juggie | mut this isnt pcsupport. |
| 19:53.12 | Juggie | but its most likely due to an older motherboard which doesnt support the larger ram. |
| 19:53.15 | Juggie | try a bios update. |
| 19:53.58 | klork | guys which soft phones work best with asterisk.... plzz help out |
| 19:54.27 | generalhan | Juggie: http://generalhan.pastebin.ca/142942 like that ? |
| 19:54.38 | obiwanmikenolte | klord: Any phone that supports sip or iax |
| 19:54.44 | generalhan | Juggie: this is my dialplan for that context |
| 19:54.56 | Juggie | something like that yes |
| 19:55.03 | generalhan | hmm ... still not working. |
| 19:55.07 | klork | ya fine but i tried some but most hang etc ... need ur experience |
| 19:55.14 | Juggie | what happens when you press * |
| 19:55.19 | generalhan | i have operator=yes in the global contect in voicemail.conf |
| 19:55.21 | generalhan | nothing |
| 19:55.40 | [TK]D-Fender | generalhan : Did you define that in each box's setup? |
| 19:55.59 | generalhan | [TK]D-Fender: im not sure what you mean |
| 19:56.04 | Juggie | generalhan, i mean what happens when you press 0 |
| 19:56.12 | [TK]D-Fender | generalhan: each box definition should have operator=yes |
| 19:56.30 | generalhan | you mean each seperate context in voicemail.conf needs that line ? |
| 19:56.55 | Juggie | it should work just defined once in general |
| 19:58.27 | generalhan | Juggie: the only button that does anything when in VM is the # |
| 19:58.27 | generalhan | and i put it in global AND in that context and still nothing |
| 19:58.30 | Juggie | i dont know then i'm busy with a million other problems atm. |
| 19:58.31 | Juggie | read the docs. |
| 19:58.36 | obiwanmikenolte | Is the call you're making actually going to the macro-incoming-calls-announced? |
| 19:58.36 | Juggie | http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+VoiceMail |
| 20:00.09 | generalhan | Juggie: i have read through the docs .. thats how i got it to where you saw it ... but i figured it out anyhow ! lol |
| 20:00.19 | [TK]D-Fender | generalhan: No, each BOX needs it. |
| 20:00.19 | obiwanmikenolte | Try making a test extension and put it in your phone's context. Make it like "exten => 666,1,Macro(incoming-calls-announced) and with whatever arguments you're passing |
| 20:00.36 | obiwanmikenolte | Fender: You mean server? |
| 20:01.06 | Juggie | generalhan, 'When using the zero '0' and star '*' it's important to note that the context you placed the application voicemail in is irrelvant, it's the context for the voicemail box that we're looking for in the dialplan for the jump to the 'a' or 'o' extention. ' |
| 20:01.07 | generalhan | there is a macro that calls that macro, and when i try from inside the office its not actually going to that context. i called on my cell and it DOES work that way ! |
| 20:01.16 | obiwanmikenolte | Right |
| 20:01.20 | obiwanmikenolte | That's your problem, then |
| 20:01.26 | generalhan | which is what i wanted to begin with ... i dont need people inside the office being transfered to someone from the VM lol |
| 20:01.32 | [TK]D-Fender | generalhan: You are very mixed up..... |
| 20:01.36 | obiwanmikenolte | Thilly General |
| 20:01.39 | generalhan | [TK]D-Fender: VERY |
| 20:02.03 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 20:02.08 | [TK]D-Fender | generalhan: You can have "o" and "a" in your macros (as well you should), and allow for variable operator's on a per-box basis. |
| 20:02.26 | *** join/#asterisk niZon (n=ilt@S0106beefd4cecc3d.wp.shawcable.net) |
| 20:02.38 | [TK]D-Fender | generalhan: As long as you return to "s" and allow your dialplan to resume after you can even continue on. |
| 20:04.12 | generalhan | so to get it to go to a different place based on which VM a caller is in wouldnt i need each of those VMBoxes in a different context? in order to specify where each go ? |
| 20:06.28 | obiwanmikenolte | I'm guessing that the answer is no, but before I go to the bugs channel: Has anyone had a problem connecting to Asterisk 1.2.10 while it's running? It's not creating /var/run/asterisk.ctl or /var/run/asterisk.pid for some reason. I tried running asterisk as root, and it still doesn't work, so it's not a permissions problem (root can write to /var/run with no problem). I've installed earlier versions without this problem, so it may be a bug, |
| 20:07.31 | hmmhesays | so my fix_nated_register is working |
| 20:07.54 | *** part/#asterisk klork (i=vny@h678631.serverkompetenz.net) |
| 20:07.58 | hmmhesays | are you sure it is running? |
| 20:08.04 | hmmhesays | ps aux | grep asterisk |
| 20:08.35 | obiwanmikenolte | Yes, it's running. |
| 20:08.44 | obiwanmikenolte | I can make calls, and it does show up on the process list |
| 20:09.05 | *** join/#asterisk DFiber (n=bwarner@65.113.208.18) |
| 20:09.05 | vosque | Is there somewher eother than the sip.conf file that you have to specify the incoming sip port? I try to change it to 5061 and "sip show settings" stays at 5060 |
| 20:09.25 | *** join/#asterisk Skarmeth (n=Skarmeth@201009005250.user.veloxzone.com.br) |
| 20:09.33 | DFiber | All: I have a simple question, when I call inbound to my asterisk server, my caller id says "asterisk" how can I change that? |
| 20:09.54 | Skarmeth | hi all |
| 20:09.56 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.221) |
| 20:09.58 | DFiber | i have my zapchannel set as as received....what gives? |
| 20:10.22 | obiwanmikenolte | DFiber: SetCallerID |
| 20:10.43 | DFiber | thanks.. |
| 20:10.51 | eKo1 | vosque: no, sip.conf is where you set that |
| 20:10.54 | Skarmeth | I was trying to limit inbound calls to a sip peer to only one, but the call-limit directive, set's both input and output call limit, this way, I lost the Transfer funcion of the phone... any way to limit only input calls? |
| 20:11.24 | eKo1 | Skarmeth: not in 1.2; only in 1.0 |
| 20:11.51 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 20:11.58 | vosque | eKo1: any idea why it might not be changing? |
| 20:12.10 | obiwanmikenolte | reload sip? |
| 20:12.29 | eKo1 | you mean 'sip reload' |
| 20:12.34 | obiwanmikenolte | Errr... vosque: sip reload in the CLI |
| 20:12.35 | vosque | obiwanmikenolte: several times, unfortunately. |
| 20:12.39 | obiwanmikenolte | Heh. |
| 20:12.46 | caio1982 | bleh, coppice isn't around for some faxing torture :) |
| 20:13.14 | eKo1 | vosque: It could be a bug. |
| 20:13.33 | vosque | eKo1: alright. |
| 20:13.35 | eKo1 | vosque: what happens when you stop and start *? |
| 20:13.41 | vosque | eKo1: the same |
| 20:13.41 | eKo1 | Does it change then? |
| 20:13.46 | vosque | eKo1: no |
| 20:13.56 | eKo1 | That is strange. |
| 20:13.57 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.21.110.Dial1.SanJose1.Level3.net) |
| 20:14.03 | eKo1 | What version of * are you using? |
| 20:14.29 | vosque | 1.2.10 |
| 20:15.30 | eKo1 | Checking... |
| 20:15.55 | hmmhesays | bah |
| 20:16.18 | Dovid | afternoon |
| 20:16.27 | Dovid | anyone have any expirience with diffretn wifi phones ? |
| 20:16.33 | Dovid | i am stuck on which one to get |
| 20:16.39 | eKo1 | vosque: where are you reading the port number from? |
| 20:16.46 | vosque | sip show settings |
| 20:17.00 | vosque | I'm also seeing if it responds on tcpdump with some nc'ing |
| 20:17.23 | hmmhesays | does anyone have a copy of the old SER admin guide? |
| 20:17.46 | eKo1 | vosque: I'm getting the same behaviour. It probably is a bug. |
| 20:17.57 | vosque | eKo1: alright, thanks. |
| 20:18.24 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:18.28 | caio1982 | does anyone here have some clue regarding this compilation error? maybe it's some obvious syntax mistake that i'm not aware? chan_sip.c related... log at http://pastebin.ca/142992 and the original patch at http://caio.ueberalles.net/asterisk_1.2.10_t38_20060817_chansip.txt |
| 20:18.50 | caio1982 | aware of* |
| 20:18.58 | eKo1 | Unless you've fiddled with chan_sip.c, there should be no mistake in it. |
| 20:19.57 | eKo1 | caio1982: blame the patch |
| 20:20.01 | caio1982 | the patch is pretty big, imho, but it has been applied nicely so far |
| 20:20.46 | Katty | hmmhesays: mew. |
| 20:21.21 | eKo1 | What does that patch do? Does it add t.38 support or something? |
| 20:21.32 | caio1982 | eKo1: so blame me :) it's based on a patch from 1.2.7... i'm almost 100% sure that's a syntax problem i dont recognize at the moment, since it applied okay, right? |
| 20:23.23 | caio1982 | right now i'm tracking gentoo' svn to check out how they applied it, when they have done it and where |
| 20:23.27 | eKo1 | caio1982: it is a syntax problem |
| 20:23.44 | caio1982 | eKo1: any hints? |
| 20:23.49 | eKo1 | download the modified chan_sip.c and look at the function where the syntax error is. |
| 20:25.31 | caio1982 | that's what i'm doing, eKo1, but seems i'm not capable to understand where it is |
| 20:25.53 | caio1982 | yet! :) |
| 20:26.17 | vosque | eKo1: could my bug be acl.c line 310? Not really a C person. |
| 20:26.18 | vosque | :q |
| 20:27.17 | eKo1 | vosque: I suggest you make a post about it in bugs.digium.com. |
| 20:27.28 | vosque | eKo1: oh, right on. Thanks. |
| 20:27.55 | hmmhesays | Hey Katty |
| 20:27.57 | hmmhesays | whats up? |
| 20:28.27 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 20:29.07 | *** part/#asterisk TripleFFFF (n=TripleFF@145-27.mc.cite.net) |
| 20:29.39 | hmmhesays | oops she left |
| 20:29.49 | wwalker | anyone here successfully used asterisk with OpenWRT? I've tried sjphone and IP500. asterisk segfaults shortly after the phone registers. |
| 20:30.09 | *** join/#asterisk champster (n=asterisk@AH.tescogroup.com) |
| 20:30.17 | wwalker | has anyone seen that failure in regular asterisk? |
| 20:30.51 | champster | Is there a trick to SendDTMF? I am trying to use it for paging zone selection on a Valcom 3 zone, and I do not get audio back immediately after the SendDTMF. |
| 20:36.33 | *** join/#asterisk NDT (n=noone@cpe-24-195-66-214.nycap.res.rr.com) |
| 20:37.04 | TrixVox | Anyone try posting to the mailing list today? My posts never went through... |
| 20:37.10 | *** join/#asterisk Flauto (n=HP_Owner@adsl-75-3-139-218.dsl.chcgil.sbcglobal.net) |
| 20:37.17 | Flauto | hi all |
| 20:37.22 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 20:38.07 | TrixVox | asterisk-users |
| 20:38.24 | Flauto | hi trix-vox |
| 20:38.33 | Flauto | are you the owner of trxtel.com |
| 20:38.38 | TrixVox | nope |
| 20:38.41 | Flauto | okay |
| 20:38.55 | TrixVox | he's in here though, forget his name |
| 20:38.59 | Flauto | i used trxtel for toll free terminations for a while |
| 20:39.12 | champster | I am using for toll free as well. |
| 20:39.12 | Flauto | the quality is very bad |
| 20:39.36 | Flauto | the other part always telling me that my voice is shaking like crazy |
| 20:39.37 | champster | I comes and goes, I am not sure if it is my connection to the net or not. |
| 20:39.49 | TrixVox | haha, i use voicepulse connect... rate is higher but quality is very good |
| 20:41.10 | Flauto | i only use it for toll free calls |
| 20:41.27 | Flauto | and now, i am using my pstn for toll free calls |
| 20:41.36 | TrixVox | oh, outbound toll free |
| 20:42.03 | Flauto | what distro is the best for asterisk |
| 20:43.39 | jbroome | ~best |
| 20:43.42 | jbot | best for what? please define what you mean by "best" Gloria Gaynor! Tina Turner! Aretha Franklin! Men without Hats! Women without Hats! Flock of Seagulls!, or fvwm! Women without clothes! |
| 20:43.45 | ki2k | Flauto: Microsoft Windows for Workgroup 3.11 |
| 20:43.59 | wwalker | jbot: Women without clothes! |
| 20:44.43 | Flauto | haha |
| 20:44.52 | Flauto | so, they are pretty much the same shit? |
| 20:44.59 | wwalker | Flauto: the one you have someone to help you with the most, or the one you known the best, or CentOS; in that order |
| 20:45.24 | wwalker | Flauto: you must be talking about windows |
| 20:45.39 | hmmhesays | how do you exit minicom |
| 20:45.46 | Flauto | walker, i have been using mandriva, but the webvmail is not working on that |
| 20:45.50 | intralanman | wwalker: ok, you said the win word, get the f out ;) |
| 20:45.51 | Qwell | hmmhesays: ctrl-q |
| 20:45.53 | Qwell | erm |
| 20:45.54 | eKo1 | hmmhesays: C-a x |
| 20:45.56 | Qwell | ctrl-a+q |
| 20:46.00 | Qwell | q quits without reset |
| 20:46.18 | wwalker | hmmhesays: C-a ? then x or q |
| 20:46.41 | Nivex | There is a small chance it might be Alt-a, depending on distro |
| 20:46.47 | hmmhesays | fc4 |
| 20:46.53 | Qwell | Nivex: Really? That's silly. |
| 20:47.06 | Nivex | The lower left corner should indicate which meta is in use |
| 20:47.13 | Nivex | Qwell: how so? |
| 20:47.14 | wwalker | Flauto: if you are comfortable with rpms, I recommend CentOS or FC4 or FC5 |
| 20:47.18 | NDT | Anyone here able to read Spanish? Heh trying to see if this damn babble fish of Alta Vista's translated some things right |
| 20:47.19 | Qwell | Nivex: just is |
| 20:47.28 | hmmhesays | thanks |
| 20:51.25 | *** join/#asterisk russellb (n=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 20:51.25 | *** mode/#asterisk [+o russellb] by ChanServ |
| 20:52.26 | *** join/#asterisk __bosse__ (n=Bo@kain.erichsen.com) |
| 20:52.47 | *** join/#asterisk IOscanner (n=IOscanne@c-67-164-154-209.hsd1.tx.comcast.net) |
| 20:52.52 | __bosse__ | anybody knows how to make asterisk sip channel ignore dtmf? If i set dtmfmode to inband it will still try to detect dtmf and relay it using the rtp code. I just want to relay the dtmf _as is_. I use g711a so there shouldn't be any probs with codec. So basically i just want asterisk to stop detecting dtmf :) |
| 20:53.17 | obiwanmikenolte | SendDTMF? |
| 20:54.19 | obiwanmikenolte | _bosse_: the SendDTMF() application might do what you want |
| 20:54.22 | obiwanmikenolte | Heh |
| 20:54.22 | __bosse__ | is used to send dtmf digits on the sip channel using the specified dtmfmode.. I wan't * to leave the audiostream and not interfere |
| 20:55.07 | champster | Is there a trick to SendDTMF? I am trying to use it for paging zone selection on a Valcom 3 zone, and I do not get audio back immediately after the SendDTMF. |
| 20:55.41 | KDan | anyone got an IAX address i can point my DID to to check whether it's my server that's wrong? Just wanna make a test call |
| 20:59.34 | IOscanner | I am setting up our own VOIP solution. I am using didx.org for the DID numbers inbound. What is a good source for outbound calling that will allow me to set the CallerID for my outbound calls for my different office locations. I want to buy one trunk that will handle all of my DID numbers for outbound calls. |
| 20:59.49 | IOscanner | Anyone know of a vendor I can use? |
| 21:00.45 | mog | ssokol, ping |
| 21:01.09 | ssokol | mog: pong |
| 21:01.10 | obiwanmikenolte | ssokol owes me a working configuration file |
| 21:01.18 | obiwanmikenolte | From the final lab |
| 21:01.22 | obiwanmikenolte | And don't think that I'll forget |
| 21:01.39 | ssokol | obiwanmikenolte: these aren't the droids your're looking for... |
| 21:01.48 | obiwanmikenolte | Move along |
| 21:01.55 | mog | should i register to come to astricon with my 0 discount code , or let digium do it later? |
| 21:01.58 | ssokol | obiwanmikenolte: they're for sale if you want them... |
| 21:02.12 | obiwanmikenolte | ssokol: I already paid |
| 21:02.19 | }btorch{ | does cdr also keeps record of incoming calls ? |
| 21:02.24 | obiwanmikenolte | Ooh, snap! |
| 21:02.33 | KDan | obiwanmikenolte: jedi mind tricks don't work on ssokol - only money |
| 21:02.38 | ssokol | mog: please register. otherwise debbie will be going insane in October trying to figure out who is registered and who isn't |
| 21:02.39 | }btorch{ | for some reason I only see records of outgoing calls |
| 21:02.49 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 21:02.52 | mog | yeah ill double check with her |
| 21:02.54 | mog | and register |
| 21:02.56 | obiwanmikenolte | Hey, even I get boarded sometimes (but I usually have to pay for that too) |
| 21:02.57 | ssokol | obiwanmikenolte: we've had them five seasons... |
| 21:03.12 | obiwanmikenolte | I was in the May class in KC |
| 21:03.17 | eKo1 | }btorch{: you're not seeing right. |
| 21:03.23 | file | mog: I registered, Lisa just sent me a semi-working code |
| 21:03.23 | KDan | anyone got an IAX address i can point my DID to to check whether it's my server that's wrong? Just wanna make a test call |
| 21:03.32 | mog | okies |
| 21:03.34 | mog | ill do that then |
| 21:03.38 | Skarmeth | ok, my last question about how to limit incoming calls was solved :) I used the device config to do so... Polycom sip.conf call.callsPerLineKey=1 |
| 21:03.44 | Skarmeth | thanks all |
| 21:04.12 | obiwanmikenolte | ssokol: the final lab had a ton of errors, and you said that you'd e-mail out a working copy |
| 21:04.25 | __bosse__ | I've dug into the source and it seems that you have to patch the sip_chan code to achieve this :( .. more specifically i think i need to change sip_chan to set a feature to disable dtmf detect alltogether.. anybody tried this? |
| 21:04.30 | ssokol | obiwanmikenolte: you're looking for the final configs for big lab with db, dnd, fwd, etc., right? |
| 21:04.31 | obiwanmikenolte | ssokol: don't let me down in front of everyone. |
| 21:04.36 | obiwanmikenolte | Yep |
| 21:04.47 | eKo1 | __bosse__: you mean chan_sip |
| 21:04.58 | ssokol | mog: thanks! lisa and I really appreciate it. |
| 21:05.11 | ssokol | obiwanmikenolte: let me see what I have here... |
| 21:05.24 | Nivex | KDan: IAX2/guest@misery.digium.com/s@default |
| 21:05.36 | hmmhesays | is true what they say, you won't give it away and I don't know what to do, to get next to you, next to you |
| 21:05.46 | KDan | Nivex: cheers, will try that now |
| 21:05.47 | hmmhesays | i've been trying all night long and I wanna get next to you |
| 21:05.55 | KDan | what's the s@default bit? |
| 21:06.01 | *** join/#asterisk |dennis| (n=dennis@200.32.215.82) |
| 21:06.04 | Nivex | exten@context |
| 21:06.17 | Nivex | I lifted that straight out of the demo extensions.conf |
| 21:06.21 | hmmhesays | tech/auth/exten@context |
| 21:06.26 | __bosse__ | _eKo1: Yep - chan_sip.c :) - around line 16185 here http://www.asterisk.org/doxygen/chan__sip_8c-source.html#l16149 |
| 21:07.00 | KDan | ok, it works |
| 21:07.06 | KDan | so it's my server that's buggered |
| 21:07.08 | KDan | cheers :-) |
| 21:07.25 | hmmhesays | LOL, thats a funneh word |
| 21:07.32 | hmmhesays | i detect a brit! |
| 21:07.41 | KDan | oh no! I've been found out! |
| 21:07.43 | Nivex | or an aussie |
| 21:07.50 | KDan | fraid not |
| 21:08.12 | KDan | actually I'm swiss, but having spent the last 9 years in england i almost qualify as one of those "brits" |
| 21:08.14 | hmmhesays | so this new buckcherry album just rocks |
| 21:08.16 | KDan | :-) |
| 21:08.29 | *** join/#asterisk ilTizio (i=foobar@adsl203-149-051.mclink.it) |
| 21:09.05 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 21:10.52 | ilTizio | hi, I'm using asterisk 1.2.10.dsfg-3 debian pkg. Is there anyone who can help me with a sip.conf register line in which the ":port" part is ignored? |
| 21:11.29 | IOscanner | Does anyone know of a provider that has IAX or SIP account for outbound calling that will allow me to change the outbound CallerID. I have DID's from didx. |
| 21:12.05 | intralanman | IOscanner: i heard voicepulse does, i haven't verified that though |
| 21:12.21 | syzygyBSD | ilTizio: post the sip.conf in question |
| 21:12.24 | IOscanner | Cool any others |
| 21:12.31 | ssokol | obiwanmikenolte: Just sent you the updated user macros |
| 21:12.32 | obiwanmikenolte | Suite! Thanks, ssokol! |
| 21:12.34 | obiwanmikenolte | Got it |
| 21:12.35 | syzygyBSD | pastebin it |
| 21:13.52 | ilTizio | syzygyBSD: register => XXXXX:YYYYY@sip.messagenet.it:5061/102 ; Italian SIP Provider |
| 21:14.03 | ilTizio | syzygyBSD: using sip debug i can see |
| 21:15.00 | ilTizio | syzygyBSD: Retransmitting #{1 .. 6} (NAT) to 212.97.59.76:5060 |
| 21:15.01 | syzygyBSD | I dont' like using the register command, normally a normal entry will work |
| 21:15.16 | ilTizio | syzygyBSD: using type=friend? |
| 21:15.41 | syzygyBSD | friend should work |
| 21:15.54 | ilTizio | syzygyBSD: I need to receive call on that account so I've to register to the provider |
| 21:16.02 | syzygyBSD | I can never remember if it is user or peer, so I normally use friend |
| 21:16.30 | syzygyBSD | friend should register too |
| 21:16.48 | ilTizio | let me try |
| 21:18.40 | *** join/#asterisk Ciber311 (n=Ciber311@user-1087e94.cable.mindspring.com) |
| 21:21.22 | ilTizio | syzygyBSD: type=friend doesn't register, I cannot receive call on that account, the port=5061 is ok in the outgoing entry, but not in registry config :( |
| 21:21.45 | syzygyBSD | pastbin your sip.conf |
| 21:21.55 | syzygyBSD | black out the passwords/usernames though |
| 21:22.23 | *** join/#asterisk crlshn (i=kvirc@operaciones3.globalnet.hn) |
| 21:30.20 | *** join/#asterisk lkj235 (n=thad@71-215-111-95.tcsn.qwest.net) |
| 21:31.17 | lkj235 | hello all, have a quick (and extremely tame/stupid --- depends how you look at ignorant folk such as myself) question in regards to dialing extensions for one of you asterisk gurus |
| 21:31.55 | }btorch{ | eKo1, nope i'm not that crazy yet .. my cdr table for my account code for example only show outgoing calls |
| 21:32.11 | }btorch{ | eKo1, did I miss a setting or something ? |
| 21:32.22 | eKo1 | incomming from where? |
| 21:33.03 | }btorch{ | eKo1, I just tested from another voip user and also from an external phone that comes in through a zap channel |
| 21:33.13 | }btorch{ | neither |
| 21:34.07 | *** join/#asterisk GerbilWrk (i=GerbilNu@65.88.144.41) |
| 21:34.22 | eKo1 | For the Zap call, you should see a record with channel = Zap/1 or what not and the number that was dialed in dst. |
| 21:34.37 | GerbilWrk | Any of yall setup a TFTP config server for the Grandstream handytones? |
| 21:35.01 | eKo1 | yuck, gs |
| 21:36.02 | GerbilWrk | ehh, i've never had an issue with them |
| 21:36.46 | TrixVox | IOscanner: yeah, connect.voicepulse.com lets you set the outbound CID number |
| 21:37.06 | GerbilWrk | I'm liking the LCR of connect.voicepulse |
| 21:37.16 | }btorch{ | eKo1, maybe i'm looking at this the wrong way ... I do an SQL query filtering by a user account code and I swear it only shows outgoing call. |
| 21:37.19 | TrixVox | GerbilWrk: oh yeah, it's pretty kickass |
| 21:37.36 | }btorch{ | of course if I just do a select * without any where statement I seea ll calls |
| 21:37.41 | }btorch{ | all calls |
| 21:37.44 | GerbilWrk | haven't used much through them though, alot of our calls are higher through them then our other carrier |
| 21:38.23 | [TK]D-Fender | lkj235: Just go ahead and ask. |
| 21:40.32 | eKo1 | }btorch{: Why don't you make a call and look at the last record that was inserted? |
| 21:42.05 | ilTizio | syzygyBSD: http://pastebin.it/1888 |
| 21:42.05 | TrixVox | GerbilWrk: who's your other carrier? |
| 21:42.05 | ilTizio | syzygyBSD: on line 35 is the registry line which is not working |
| 21:42.12 | GerbilWrk | Voxee |
| 21:42.36 | TrixVox | GerbilWrk: so you mean lower int'l rates? |
| 21:42.44 | GerbilWrk | US |
| 21:42.58 | syzygyBSD | ilTizio: which one below is it? |
| 21:43.02 | syzygyBSD | or is it there |
| 21:43.05 | }btorch{ | eKo1, yes that works fine ...like a said maybe I was looking for something the wrong way.. I thought all outgoing/incoming calls would be recorded under the user accountcode |
| 21:43.14 | }btorch{ | but it doesn't seem to work that way |
| 21:43.48 | GerbilWrk | TrixVox, we've used like $1.50 of Voicepulse, and like $4 of Voxee since last Monday or so |
| 21:43.53 | ilTizio | it is there: 35th line |
| 21:43.53 | TrixVox | GerbilWrk: what i do in that case is send all calls where voicepulse is 1.1c or higher through voxee (or similar flat rate provider) and all the rest through voicepulse |
| 21:44.05 | syzygyBSD | ilTizio: no, the other entries |
| 21:44.06 | GerbilWrk | yeah, that's what we are doing |
| 21:44.10 | GerbilWrk | that's why I like their LCR |
| 21:44.21 | eKo1 | }btorch{: that wouldn't make any sense. |
| 21:44.21 | syzygyBSD | here, comment out line 35, change line84 to friend |
| 21:44.28 | TrixVox | GerbilWrk: yeah, depends on who you're calling, i have some customers that do a lot of mobile calls, which means voicepulse is more expensive |
| 21:44.44 | ilTizio | syzygyBSD: I did and I could call but I could not receive call |
| 21:44.44 | }btorch{ | oh well life doesn't make sense sometimes :-) |
| 21:44.53 | TrixVox | but for landline calls, they are super cheap, most of our calls complete for $0.007 or less through them |
| 21:44.58 | syzygyBSD | did you change it to friend? |
| 21:45.06 | syzygyBSD | what was the status of the line? |
| 21:45.15 | GerbilWrk | yeah, they aren't too bad, either way, we never pay more then 1.1c :) |
| 21:45.34 | hmmhesays | interesting mr bond |
| 21:45.36 | TrixVox | yeah, using both together you're probably averaging less than 1c, which is nice |
| 21:45.48 | hmmhesays | dearly beloved are you listening |
| 21:45.56 | hmmhesays | I can't remember a word that you were saying |
| 21:46.08 | hmmhesays | are we demented or am i disturbed |
| 21:46.26 | ilTizio | with type=friend I cannot see anything about messagenet in 'sip show registry' |
| 21:46.28 | hmmhesays | the space thats in between insane and insecure, oh therapy can you please fill the void, am I retarded or am I just overjoyed |
| 21:46.51 | hmmhesays | ilTizio: could be cause setting a type has nothing to do with what shows up in sip show registry |
| 21:47.04 | ilTizio | and 'sip show peer messagenet-mi-out' show status unmonitored |
| 21:47.24 | hmmhesays | and will always do so if you have no qualify=yes |
| 21:47.38 | ilTizio | hmmhesays: yes I know, but I cannot understand why the :port in registry line is ignored |
| 21:47.49 | hmmhesays | I highly doubt it is |
| 21:47.58 | ilTizio | but it is |
| 21:48.02 | hmmhesays | unlikely |
| 21:48.13 | ilTizio | I cannot understand but ... http://pastebin.it/1888 |
| 21:48.28 | hmmhesays | do you have the time to listen to me whine.... take it #asterisk |
| 21:48.38 | ilTizio | is my sip.conf and with sip debug i can see the following: |
| 21:48.41 | [hC] | any of you guys use or know of someone who has implemented door locks controlled by asterisk? |
| 21:48.49 | [hC] | like a door buzzer? |
| 21:49.04 | KDan | Aug 21 22:47:59 NOTICE[2147]: chan_iax2.c:6899 socket_read: Rejected connect attempt from 213.230.216.67, request 's@iax' does not exist <<< Why does it say this "s@iax" stuff when the address i specified was guest@myserver ?? |
| 21:49.27 | file | KDan: you specified guest as the username, and no extension - so it used 's' |
| 21:49.28 | hmmhesays | it wouldn't be hard being you can do system calls with asterisk |
| 21:49.35 | KDan | aaah |
| 21:49.40 | KDan | thanks |
| 21:49.46 | file | you're welcome |
| 21:50.01 | hmmhesays | grasping to controllllll so I better hold ooooooon |
| 21:50.11 | syzygyBSD | ilTizio: did you comment out register when you put in friend/ |
| 21:50.34 | file | hmmhesays: poke! |
| 21:50.38 | ilTizio | yes i did |
| 21:50.38 | hmmhesays | hola file |
| 21:50.42 | file | hola! |
| 21:50.46 | hmmhesays | am I just paranoid or am I just stoooooned |
| 21:50.49 | ilTizio | hmmhesays: Reliably Transmitting (NAT) to 212.97.59.76:5060: |
| 21:50.56 | ilTizio | hmmhesays: REGISTER sip:sip.messagenet.it SIP/2.0 |
| 21:50.56 | file | hmmhesays: both |
| 21:51.20 | hmmhesays | is 212... the ip sip.messagenet.it resolves to? |
| 21:51.40 | ilTizio | hmmhesays: yes it is |
| 21:51.54 | hmmhesays | do you get that when you are registering or when you send a call to them |
| 21:52.02 | [TK]D-Fender | hmmhesays: See.... Basket Case = better opener! :D |
| 21:52.07 | num000 | pango awaik? |
| 21:52.16 | ilTizio | when registering |
| 21:52.19 | hmmhesays | [TK]D-Fender: still hurts to play |
| 21:52.43 | hmmhesays | my wrist is still messed up from dumping my bike last month |
| 21:52.44 | lkj235 | [TK]D-Fender: Thank you for your response, I apologize for the delay, as I had to take a call. Anyways, my question is: How do you transfer a call from within the dialplan or via AGI (I'm using PHPAGI for the most part) to an extension?: Goto(), Dial(), or some other command? |
| 21:52.47 | [TK]D-Fender | hmmhesays: HURTS!? Really shouldn't..... |
| 21:52.58 | hmmhesays | i messed it up in the bike accident last month |
| 21:53.11 | hmmhesays | 3 minutes of power chords gets painful |
| 21:53.16 | [TK]D-Fender | lkj235: Well transferring a call on a phone depends on the phone... |
| 21:53.27 | [TK]D-Fender | hmmhesays: Ok, I guess there's no easy way outta that... |
| 21:53.39 | hmmhesays | just some more time, it is getting better |
| 21:53.40 | lkj235 | well I'm using a pap2 and soon to be using a utstarcomm f3000 as well |
| 21:53.56 | lkj235 | so there's not a whole lot of control since it's (well, at least the pap2t) just going to "dumb" phones |
| 21:54.14 | [TK]D-Fender | lkj235: Well you use hookflash + maby * codes for the PAP2 depending, and the UTC should have a transfer soft-key somewhere on it. |
| 21:54.49 | hmmhesays | time to go home |
| 21:54.57 | [TK]D-Fender | lkj235: Nothing * should have to think about... its up to the end point, and THEN if the endpoint is exceedingly stupid, should offer the options through straight DTMF (a bad practice) |
| 21:55.05 | ilTizio | hmmhesays: thx for your time, bye |
| 21:55.07 | [TK]D-Fender | maybe* |
| 21:55.14 | file | [TK]D-Fender: stupid is as stupid does |
| 21:55.39 | [TK]D-Fender | file: He's Gump, he's Gump, is he BRAIN-DEAD? ....... |
| 21:55.50 | lkj235 | k that part is understandable, but I'm saying if I don't answer after say 30 seconds, then how do I get it to route to an extension within my context where I have my voicemail setup with agi |
| 21:56.41 | lkj235 | [TK]D-Fender: I'm using queues so that it will fail after 30 seconds and ringing the 2 devices (right now I'm just making use of the pap2's 2 sip ports until I can throw the f3000 in the mix) |
| 21:56.51 | mog | ssokol, i cant pay an ammount of 0.00 with paypal :( |
| 21:57.09 | file | mog: email Lisa, she can take care of it... she did for mine |
| 21:57.15 | mog | im gonna use my cc but i dont like giving it out |
| 21:57.23 | TrixVox | ssokol: how many people going to astricon? |
| 21:57.35 | file | Lisa rocks! |
| 21:57.42 | [TK]D-Fender | lkj235: Queues sounds like massive overkill.. what do you really want to do? |
| 21:57.57 | lkj235 | [TK]D-Fender: and I wanted to do my own voicemail system via AGI because I'm needing a ton more flexibility for doing some 'abstract' voicemail stuff than what I'm able to use with the [extremely nice] * voicemail system |
| 21:58.26 | Juggie | mog, get a CC for just online |
| 21:58.29 | lkj235 | I just want it to try ringing my devices for 30 seconds or so and then dump into AGI that I have sitting on a seperate extension |
| 21:58.29 | [TK]D-Fender | lkj235: How do queue's factor into this, let alone "transfers"? |
| 21:58.30 | Juggie | if your not trustworthy |
| 21:58.40 | mog | i do do that now |
| 21:58.51 | Juggie | i think some companies have pre-paid visa avail too |
| 21:58.58 | Juggie | i just trust... perhaps i'm too trusting |
| 21:59.01 | mog | hmm wont let me do it with cc either |
| 21:59.05 | [TK]D-Fender | lkj235: You just dial for 30s and check the dialstatus variable and choose where to go from there..... |
| 21:59.05 | mog | im just paranoid |
| 21:59.15 | lkj235 | [TK]D-Fender: transfers was probably the wrong word to use, but basically I just want my AGI sitting on an extension so I can dial into it, or have the call "transferred" to the AGI's extension after 30 seconds. Teh que is so it will try ringing my f3000 and pap2 at the same time |
| 21:59.34 | file | mog: you're also mog |
| 21:59.35 | lkj235 | [TK]D-Fender: (that way if I'm up @ college or work or whatever with my f3000 then it will ring at the same time as my house) |
| 22:00.17 | [TK]D-Fender | lkj235: Do dial 2 at the same time you just do something like this : Dial(SIP/pap2@SIP/f3000,30) |
| 22:00.18 | Juggie | mog, your not liable for it why worry. |
| 22:00.26 | file | & |
| 22:00.32 | [TK]D-Fender | lkj235: Queue = silly |
| 22:00.36 | file | SIP/pap2&SIP/f3000 |
| 22:00.54 | mog | yeah i know |
| 22:00.59 | mog | but i have had to get money back |
| 22:01.05 | mog | due to a fradulent company |
| 22:01.08 | mog | took 3 months |
| 22:01.13 | mog | even though i did everything right |
| 22:01.23 | mog | but i wasnt liable which was nice |
| 22:01.59 | *** join/#asterisk roving_prole (n=Harper@72-254-127-241.client.stsn.net) |
| 22:02.02 | [TK]D-Fender | lkj235: Do dial 2 at the same time you just do something like this : Dial(SIP/pap2&SIP/f3000,30) |
| 22:02.09 | Juggie | yah i guess the waiting sucks |
| 22:02.09 | [TK]D-Fender | yes... minor typo. |
| 22:02.11 | Juggie | thats the worst part. |
| 22:02.18 | Juggie | usually its faster |
| 22:02.38 | [TK]D-Fender | ok, gotta jet.. back later maybe |
| 22:02.41 | Juggie | its never happened to me but it happened to a friend of mine and he has his money back in like 2 days |
| 22:02.56 | mog | yeah that is what is supposed to happen |
| 22:03.01 | *** part/#asterisk obiwanmikenolte (n=obiwanmi@mail.efc-intl.com) |
| 22:03.06 | mog | although with me the company i did it with went dark |
| 22:03.10 | mog | fell of face of planet |
| 22:03.21 | mog | and when visa called they claimed they had shipped it |
| 22:03.22 | TrixVox | what company? |
| 22:03.26 | mog | and then they fell off |
| 22:03.32 | mog | some company i found of pricewatch |
| 22:03.37 | mog | it happened in highschool |
| 22:03.40 | lkj235 | [TK]D-Fender: but I also want to have "ring music" going which is why I use the ques |
| 22:03.42 | mog | i bought a graphics card |
| 22:03.43 | lkj235 | *queues |
| 22:03.48 | mog | and the deal was too good to be true |
| 22:03.50 | mog | turned out it was |
| 22:04.00 | lkj235 | so when people call me right now, they get to hear Irish Beer Drinking songs while my phones rings. :-D |
| 22:04.23 | syzygyBSD | when they claimed they shipped it.. ask for a tracking number |
| 22:04.31 | mog | yeah its what i did |
| 22:04.34 | mog | we had big fight |
| 22:04.35 | lkj235 | hence why I use queues instead of just Dial() 8-) |
| 22:04.41 | mog | cc sided with me |
| 22:04.49 | mog | just took long time |
| 22:04.52 | lkj235 | but my question is, how do I sit there and then grab it after 30 seconds and dump it into the extension where the AGI is at |
| 22:04.57 | syzygyBSD | lkj235: you can do that with dial too |
| 22:05.07 | mog | what syzygyBSD said |
| 22:05.11 | mog | its like option m |
| 22:05.22 | syzygyBSD | yup |
| 22:05.28 | *** part/#asterisk Alric (n=nbowyer@masq.hyperusa.com) |
| 22:05.29 | DaKalle | appro options in the dial command: |
| 22:05.47 | syzygyBSD | but queues also have a timeout if I recall correctly |
| 22:05.47 | DaKalle | is it possible that the o option does not what it is said to do |
| 22:05.50 | DaKalle | ? |
| 22:05.50 | lkj235 | in which case I have that as extension *123 and using the same code to handle each (and just parsing the hell out of the variables when * dumps it into the agi |
| 22:06.59 | lkj235 | syzygyBSD: How do you do it with dial()? And if you can ring multiple at the same time and have random songs being played while the caller is waiting, then what's the point of queues having strategy=ringall ? |
| 22:07.15 | *** join/#asterisk dasenjo (n=dasenjo@208.195.215.99) |
| 22:07.42 | syzygyBSD | lkj235: because queues will handle multiple calls; more calls the phones |
| 22:08.00 | DaKalle | because my asterisk doesent submit the caller id from the caller, but from the extension which is called |
| 22:08.00 | syzygyBSD | then* |
| 22:08.11 | Dovid | anyone know about polycom and MWI ? |
| 22:08.23 | mog | what about it |
| 22:08.50 | DaKalle | and there is no difference between doing the dial command with or without the o option |
| 22:08.58 | syzygyBSD | lkj235: it is the music on hold option for dial, then the random songs is the same random anything with MoH |
| 22:09.32 | file | DaKalle: what exactly is your issue? |
| 22:10.26 | DaKalle | i want my sip phones to ring in different ringtones, one for internal calls and one from outside my company |
| 22:11.09 | KDan | anyone know what codec(s) skype uses? |
| 22:11.13 | Dovid | mog: cant seem to get the light to blink when there is a vm waiting |
| 22:11.19 | DaKalle | and when asterisk lets an extension ring, the extension always says it is called by himself |
| 22:11.30 | mog | ? |
| 22:11.38 | file | DaKalle: what is your exact Asterisk version, and dial line? |
| 22:11.39 | mog | i can tell you it works |
| 22:11.45 | mog | you must have it misconfigured Dovid |
| 22:12.12 | lkj235 | syzygyBSD: hmmmmm. So what happens if I use dial() and I pick up a line and someone calls. Does it ring the other phone (the f3000) or just skip over and then goto (my original question :-)) of bouncing to the extension? |
| 22:12.13 | Dovid | mog: i know the question is what. a friend of mine did it b4 but just wit making changes on the asterisk side |
| 22:12.23 | lkj235 | syzygyBSD: (in which case, how do you do that? goto() ?) |
| 22:12.35 | DaKalle | i don't exactly know the version, because at my company, there is an internet breakdown |
| 22:12.35 | mog | it should work if you tell it in sip.conf that phone is attached to vm |
| 22:12.46 | DaKalle | but it must be som early 1.2 version |
| 22:12.48 | Dovid | mog: using real time |
| 22:12.57 | KDan | anyone know what codec(s) skype uses? |
| 22:12.58 | syzygyBSD | goto(context,extension,priority) |
| 22:13.01 | Dovid | mog: under mailbox i have 12@context |
| 22:13.12 | lkj235 | mog: you talknig to me (ref vm and phone) or someone else? |
| 22:13.19 | syzygyBSD | and it depends on how many calls that phone is allowed |
| 22:13.51 | lkj235 | syzygyBSD: so if my context is [inbound] then goto(inbound, 2)? I'm just using n for the priority since it's easier to cut and paste in the order I want than to renumber everything |
| 22:13.59 | Dovid | 12 being the box number but it still wont work |
| 22:14.38 | DaKalle | and the dial command must be somewhat like Dial(SIP/21 & SIP/22..., 30, tTo) |
| 22:14.56 | *** part/#asterisk num000 (n=numerobi@e177180054.adsl.alicedsl.de) |
| 22:16.07 | Dovid | mog: ? |
| 22:16.24 | mog | yes |
| 22:16.31 | syzygyBSD | lkj235: goto([[context,]extension,]priority) |
| 22:16.32 | mog | sorry got distracted |
| 22:16.38 | Dovid | tis ok |
| 22:16.46 | *** join/#asterisk IlTizio (i=foobar@adsl203-149-051.mclink.it) |
| 22:16.57 | Dovid | under mail box i have exten@context and still dosent work |
| 22:17.02 | Dovid | mailbox* |
| 22:17.07 | lkj235 | syzygyBSD: so if my context is [inbound] then goto(inbound, 2)? |
| 22:17.13 | syzygyBSD | no |
| 22:17.35 | lkj235 | so goto(inbound, 2, 1)? |
| 22:17.36 | syzygyBSD | look at what is required |
| 22:17.48 | lkj235 | yeah I saw, but wasn't sure if there was a way to skip the priority part |
| 22:17.58 | syzygyBSD | that would work to go to exten 2, priority |
| 22:17.59 | syzygyBSD | 1 |
| 22:18.12 | lkj235 | syzygyBSD: (in which case I'm very appreciative btw, please do not think that I'm trying to waste your time. I'm learning) |
| 22:18.19 | KDan | what's the dialplan or AGI function to read digits being typed (e.g. what would you use to read a PIN being typed in? receive_char??) |
| 22:18.40 | syzygyBSD | agi is wait_for_digit |
| 22:18.49 | KDan | cheers! |
| 22:18.59 | syzygyBSD | dialplan is just done through the extension |
| 22:19.11 | syzygyBSD | <PROTECTED> |
| 22:19.53 | IlTizio | /last KDan |
| 22:19.58 | *** part/#asterisk IlTizio (i=foobar@adsl203-149-051.mclink.it) |
| 22:20.01 | lkj235 | syzygyBSD: k I think I got it for now then, is there a way to use Dial() to goto an extension within as well, or is that the whole point of goto()? |
| 22:20.20 | syzygyBSD | what do you mean? |
| 22:20.52 | syzygyBSD | dial dials a phone, goto goes to.... |
| 22:22.33 | lkj235 | syzygyBSD: I'm not articulating this very well, and I apologize, I'm wondering what the best way is for having it dial an extension "locally"?: Goto() or Dial()? From what I understand from you helping me, I'm guessing Goto()? |
| 22:22.34 | DaKalle | file ? |
| 22:23.05 | file | DaKalle: I need the exact version and exact dial string, plus information about what kind of technology the call is coming in on |
| 22:23.52 | syzygyBSD | lkj235: lol.. well that depends on what you mean by extension |
| 22:24.28 | lkj235 | syzygyBSD: well 123 is the extension for my agi'd voicemail system (once again for anyone who's going to respond that * has voicemail, yes, I know, but it doesn't suite my *exact* needs :-)) |
| 22:24.37 | syzygyBSD | Dial if you want to go to a sip/zap phone. Goto if you just want to go to another place in the extensions.conf |
| 22:24.44 | lkj235 | er *123 |
| 22:24.54 | syzygyBSD | yes goto |
| 22:25.11 | syzygyBSD | goto(*123,1) |
| 22:25.33 | lkj235 | syzygyBSD: excellent! *THANK YOU* so much for your help! |
| 22:25.56 | syzygyBSD | np |
| 22:27.28 | DaKalle | ok, i will find that out, i will ask her a second time in the next days and i hope the internet connection problem will be solved |
| 22:27.45 | lkj235 | ok all, as always, thank y'all so much for all of y'alls help (especially syzygyBSD for spending so much time on helping me) and I hope everyone has a wonderful afternoon/evening :-) |
| 22:28.15 | syzygyBSD | I need a new name, I haven't been on bsd for 4 years |
| 22:28.37 | blitzrage | heh |
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| 22:29.59 | Corydon-w | s/BSD/LNX/ |
| 22:31.49 | blitzrage | s/LNX/WIN |
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| 22:44.56 | mtgh | If I am in the us and want to call 46-31-450..... What do I need to dial first? |
| 22:45.21 | Corydon-w | 011 |
| 22:45.52 | Corydon-w | It's in the front of the phonebook, btw |
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| 23:08.15 | syzygyBSD | hah, who still has phone books? |
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| 23:12.39 | ki2k | anyone know of ways to decrease lag w/ conference calls? |
| 23:14.23 | Corydon-w | Who here has successfully gotten the phone company to stop delivering them? |
| 23:14.33 | Corydon-w | I haven't used one in probably 5 years |
| 23:14.39 | ki2k | me neither |
| 23:14.43 | ki2k | i hate them |
| 23:15.19 | jbroome | they get dropped at our common mailbox and i leave ours |
| 23:15.27 | Corydon-w | Oh, wait. I think I used one when my A/C was malfunctioning and I had to shut down my computers due to the heat |
| 23:15.47 | jbroome | They're good to have around if you need to hit someone and not leave a mark |
| 23:15.59 | Corydon-w | I used one to find an AC repairman |
| 23:16.11 | jbroome | Yeah, that'll work too |
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| 23:55.05 | mountainm2k | PRI line -- outbound caller ID (to cell phones) shows only the extension... What's the best way to change that? |
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