| 00:00.07 | hhkahya | sorry for that unknow action |
| 00:01.18 | *** join/#asterisk TheDavidFactor (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 00:03.28 | netpro25_ | hhkahya: lol |
| 00:03.32 | netpro25_ | kinda dull in here |
| 00:03.48 | netpro25_ | people must hang out in here while they are at work |
| 00:04.35 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-148.cablep.bezeqint.net) |
| 00:05.27 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 00:06.28 | bmoraca_work | indeed |
| 00:12.36 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 00:18.36 | *** join/#asterisk voxter (n=voxter@S010600090f53ea17.vc.shawcable.net) |
| 00:19.26 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 00:22.29 | *** join/#asterisk Whtsup (n=sssi@WimaxUser3715-103.wateen.net) |
| 00:22.41 | Whtsup | how can i limit the incoming call in iax |
| 00:27.45 | Whtsup | hi |
| 00:27.49 | Whtsup | ? |
| 00:29.37 | p3nguin | Limit it to what? |
| 00:31.18 | Whtsup | like |
| 00:31.27 | Whtsup | we can use call-limit in sip.conf |
| 00:31.39 | Whtsup | i want to make this call limit in iax.con |
| 00:31.44 | Whtsup | iax.conf |
| 00:36.22 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 00:37.49 | p3nguin | see maxcallnumbers |
| 00:38.09 | Whtsup | where |
| 00:38.44 | p3nguin | iax.conf |
| 00:38.50 | Whtsup | ok |
| 00:38.52 | Whtsup | lemme check |
| 00:39.35 | p3nguin | And whoever wrote the comments in that section of the example file needs to learn the difference between "its" and "it's." |
| 00:40.45 | p3nguin | Is there a "proof reading" section on the bug tracker? |
| 00:41.26 | freetown2 | grammar nazi |
| 00:41.35 | p3nguin | Learn English and there won't be a problem. |
| 00:41.45 | p3nguin | How do I find out who wrote and commented the example configs? |
| 00:43.10 | freetown2 | heh, i don't think it is a problem with learning English. I mean, if unis in Britain, Australia and the States all complain about the command of English of their local students in writing English.... |
| 00:43.47 | freetown2 | i heard that Britain reintroduced grammar drills in 99 so maybe we will see some changes :P |
| 00:44.05 | p3nguin | Someone needs to reteach both possessive pronouns and contractions. |
| 00:44.48 | freetown2 | ha! I hear that finding that someone can be problematic |
| 00:45.15 | p3nguin | I would LOVE to teach it. The real issue is getting dense people to understand and comply. |
| 00:45.47 | freetown2 | i don't know about dense...lazy more like it :P |
| 00:45.50 | *** join/#asterisk RypPn (n=TuMbL@rosscom.co.uk) |
| 00:45.55 | Whtsup | there is no maxcallnumbers in iax.conf |
| 00:46.05 | p3nguin | What version are you using? |
| 00:46.09 | Whtsup | 1.4 |
| 00:46.13 | p3nguin | .what? |
| 00:46.26 | freetown2 | 1.4.? |
| 00:46.27 | Whtsup | asterisk 1.4 |
| 00:46.37 | Whtsup | 1.4.22 |
| 00:46.38 | p3nguin | 1.4.WHAT? |
| 00:46.40 | p3nguin | okay |
| 00:46.54 | p3nguin | Then I believe you when you said it isn't in there. |
| 00:47.07 | Whtsup | hmmm |
| 00:47.22 | Whtsup | wht shud i do any solution for that |
| 00:47.23 | p3nguin | I was expecting 1.4.<something-current> |
| 00:47.24 | freetown2 | upgrade is what he means |
| 00:47.42 | freetown2 | what other solution can you think of? |
| 00:47.46 | p3nguin | Well, at least google the option and see if it fits your need before upgrading. |
| 00:48.07 | Whtsup | ok i will try this |
| 00:48.39 | Whtsup | buh its weird we can limit call in sip it shud be in iax.conf too |
| 00:48.55 | p3nguin | I agree. |
| 00:49.31 | freetown2 | waits for grammar nazi mode |
| 00:50.14 | p3nguin | /: |
| 00:50.42 | freetown2 | anybody tried a wireless-n snom 820 yet? :-D |
| 00:50.50 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 00:50.51 | Nugget | http://nugget.livejournal.com/137834.html <-- grammar |
| 00:51.21 | *** join/#asterisk coppice (n=chatzill@25.176.64.202.dyn.pacific.net.hk) |
| 00:51.24 | citywok | i'm having trouble getting dahdi to work properly for meetme support in *1.6.1, it loads, dahdi show channels shows the pseudo device in service, but when i make a meetme conf call i get "Unable to open pseudo device" |
| 00:51.29 | freetown2 | or have comments about the snom 820 wireless-n or not |
| 00:51.30 | p3nguin | It's hard to discern when people simply leave out apostrophes completely when typing, but when they erroneously add one in in a possessive pronoun, that sticks out like a sore thumb. |
| 00:54.23 | p3nguin | If you don't put an apostrophe in "his" or "hers," what makes a person think one belongs in "its?" It makes no sense to me. |
| 00:55.09 | freetown2 | they probably never really bother to learn the difference between it's and its. That it's name :P |
| 00:56.02 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 00:56.05 | freetown2 | at least when I was in school, I remember me pals not liking grammar at all and thinking it's just one big bother |
| 00:57.19 | carrar | grammar isn't for everyone |
| 00:58.03 | coppice | i thought everyone liked grandma |
| 00:58.30 | carrar | only if she's a bakin grandma |
| 00:58.39 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 01:01.23 | freetown2 | wow [TK]D-Fender back after 3.5 hours |
| 01:01.37 | p3nguin | freetown2: Yeah, I can see how it's a bother to learn grammar and its usage, but considering that we all use words to communicate rather than grunts and moans, it does seem rather important. |
| 01:02.11 | leifmadsen | p3nguin++ |
| 01:02.35 | coppice | many people only deviate from grunts and moans on rare occasions |
| 01:02.44 | p3nguin | yeah, you're right |
| 01:02.55 | freetown2 | p3nguin, definitely agree - the problem is when you have idiot officials in the Education Burean who tell schools that grammar drills are not important and point to Britain, Australia and the States as examples |
| 01:03.06 | coppice | of course, they have the advantage of speaking a universal language |
| 01:03.10 | freetown2 | s/Burean/Bureau/ |
| 01:03.16 | freetown2 | ? |
| 01:03.24 | freetown2 | <PROTECTED> |
| 01:03.31 | freetown2 | oh |
| 01:03.41 | p3nguin | Ah, you're not familiar with infobot? |
| 01:03.57 | freetown2 | nope, just started hanging around here :P |
| 01:04.01 | coppice | The new Makita Grammar Drill |
| 01:04.10 | freetown2 | i know the tilde gets it started though |
| 01:04.50 | coppice | are grammar drills at all effective? |
| 01:04.54 | freetown2 | had a spat with a vice-principal at a school that was supposed to be one of the better ones about their English language teaching approach and policy |
| 01:04.56 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 01:05.25 | freetown2 | coppice, the Brit's have reintroduced it in 1999 i hear. I personally will say that they are |
| 01:07.35 | coppice | I was at school in the UK in the 60s when the system was trying to destroy english teaching, and those of us lucky enough to have guerilla warfare teachers were blessed. |
| 01:09.16 | freetown2 | sorry, don't know anything about that. Not born yet and I was born in HK but I grew up in Sierra Leone. |
| 01:09.54 | *** join/#asterisk mchou (n=quassel@unaffiliated/mchou) |
| 01:11.15 | coppice | english teaching here in HK is pretty poor |
| 01:11.51 | freetown2 | that's an UNDERSTATEMENT. where are you in HK? |
| 01:12.53 | coppice | I'm right here vvvvvvv |
| 01:13.10 | freetown2 | :D |
| 01:13.36 | coppice | at this moment I am in Discovery Bay |
| 01:14.12 | carrar | coppice, hows that internet censorship |
| 01:14.17 | freetown2 | you do not happen to be in the voip business here? |
| 01:14.18 | carrar | heh |
| 01:14.27 | freetown2 | carrar, what censorship? |
| 01:14.35 | carrar | You're in China right? |
| 01:14.40 | freetown2 | this is HK...them rules have not yet reached us :P |
| 01:14.51 | coppice | what internet censorship? all those western companies that block anything from an Asian internet address? |
| 01:14.53 | freetown2 | maybe another five years.... |
| 01:15.18 | coppice | carrar: with that level of understanding of geography you must be american |
| 01:15.24 | carrar | and blocking China for good reason |
| 01:15.45 | coppice | its called censorship |
| 01:15.59 | freetown2 | coppice, that Miss USA thing go around a lot eh? |
| 01:16.08 | freetown2 | s/go/got/ |
| 01:16.13 | p3nguin | When a call comes in, MixMonitor() is started before Dial()ing my phone. Once I answer the phone, both sides of the call are recorded as they should be. If I transfer the call to another phone, MixMonitor() dies once the other phone has answered. Is there any way to continue recording the call through a transfer? |
| 01:16.20 | drmessano | Blocking China for good reason? |
| 01:16.31 | drmessano | Without the chinese, there wouldn't be a Wal-Mart |
| 01:16.33 | freetown2 | spam, blah blah |
| 01:16.54 | coppice | drmessano: was that a pro or anti sentiment? |
| 01:17.00 | carrar | haha |
| 01:17.13 | freetown2 | i have this spam that was relayed through the Ministry of Foreign Affairs mail server. Keep the blocks up! |
| 01:17.48 | *** join/#asterisk slima (i=slima@unaffiliated/slima) |
| 01:18.01 | drmessano | Pro, for sure.. I have like 10 $2 Bluetooth dongles thanks to the chinese. Left to american retail, I would be out like $300 |
| 01:18.06 | *** join/#asterisk chendy (n=chatzill@119.122.245.85) |
| 01:19.17 | drmessano | If not for the chinese, we would still be trading seal fat for bear scrotum bean bags and firewood at Flo's general store |
| 01:19.23 | drmessano | Now I can buy them on ebay for $3 |
| 01:19.39 | drmessano | or Walmart for $4.50 |
| 01:20.02 | drmessano | Which is good, because I am fresh out of seal fat |
| 01:21.29 | freetown2 | if not for the chinese government make that. the poor saps that have to do the grunt work sure are not thinking about your benefits... |
| 01:22.33 | coppice | thank all governments. globalisation means no government has any power, and they all chase big companies with sweet financing deals |
| 01:23.36 | freetown2 | waits to see what happens in the spat between Google and the Chinese government |
| 01:23.43 | *** join/#asterisk staffmember (n=jt@D8FF7105.cst.lightpath.net) |
| 01:23.46 | carrar | Google should pull out |
| 01:24.00 | freetown2 | oh yeah |
| 01:24.08 | drmessano | Google everything looks like something you would buy from China on eBay.. they should be in bed together |
| 01:24.15 | drmessano | Hell, google even sounds like a knock off brand |
| 01:24.20 | carrar | censorship is something google should not be a part of |
| 01:24.26 | freetown2 | I liked it when the EU showed some backbone in dealing with Microsoft |
| 01:24.47 | staffmember | anyone know where i can find some docs on interfacing an asterisk system with a a nortel system |
| 01:24.50 | drmessano | The EU was retarded in how they handled Microsoft |
| 01:24.52 | staffmember | all the docs i can find |
| 01:24.58 | staffmember | any help would be appreciated |
| 01:25.05 | hardwire | carrar: vs abstain? |
| 01:25.17 | freetown2 | drmessano, oh you mean they should have been harder? Sure! |
| 01:25.22 | carrar | Let China sensor their own people |
| 01:25.47 | carrar | most smart people in China know how to get around their sensorship anyways |
| 01:25.50 | hardwire | drmessano: need some more seal fat? |
| 01:25.53 | *** join/#asterisk Trido (n=Trido@easter27.lnk.telstra.net) |
| 01:26.04 | freetown2 | staffmember, what choices are there in interfacing with the nortel? |
| 01:26.05 | hardwire | I could probably ask around a little. |
| 01:26.28 | drmessano | The N versions of XP show the EU was just out to put a notch on a bedpost, and could care less about setting a precedent |
| 01:26.29 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 01:26.37 | staffmember | have an AEX800 with 4port fxo/fxs.. nortel system has call pilot 100 and compact ICS |
| 01:27.35 | freetown2 | drmessano, oh, did not look at that, was more interestign the release of the cifs protocol and other stuff related to getting samba to take over windows server roles :P |
| 01:27.49 | drmessano | Its like making your kid eat more veggies or forcing him to shove an apple down his throat RIGHT NOW DAMNIT BECAUSE |
| 01:28.02 | staffmember | freetown2: do you know where i can find some info for this setup ? |
| 01:28.09 | staffmember | ior just any info in general that can help me out |
| 01:28.51 | coppice | when regulators take serious action against you its a strong sign you aren't being sweet enough to them |
| 01:28.52 | freetown2 | staffmember, well, i don't know what you want to achieve. what do you mean by interfacing? |
| 01:29.36 | freetown2 | coppice, the EU is also corrupt!? and I thought it was just every other continent :P |
| 01:30.04 | freetown2 | staffmember, route sip calls from asterisk to nortel, etc |
| 01:30.26 | staffmember | im not sure that i can route calls to the nortel compact ics.. i just want to be able to have the 2 systems talk to each others extensions |
| 01:30.28 | *** join/#asterisk pawz (n=pawz@ppp118-208-106-20.lns20.bne4.internode.on.net) |
| 01:30.46 | drmessano | Honestly, "forcing" Microsoft to release some protocol specs was almost useless. Sure some of it was used elsewhere, but no one with the power to enact real change is going to implement a product using a Microsoft protocol to sway market share. They will ofter better alternatives |
| 01:31.02 | *** part/#asterisk Tech_Travis (n=Travis@mail.techglia.com) |
| 01:32.18 | freetown2 | drmessano, yeah but when there are no alternatives to software made for Windows...it is just better to get a more stable and secure backend for the moemnt |
| 01:33.16 | drmessano | Say we do have a fully functioning Samba AD Domain Controller.. If my business is large enough that I need management with AD I am probably just going to buy a real Windows server license |
| 01:34.35 | freetown2 | why? it's not jsut a windows server license. you'd need cal licenses too. nevermind the random checks from Microsoft you might get |
| 01:34.52 | carrar | Whats windows |
| 01:34.57 | carrar | House windows? |
| 01:35.31 | freetown2 | carrar, it's not worth knowing, don't worry about it. if you have to, then poor you (me) |
| 01:35.32 | p3nguin | Car windows. |
| 01:35.46 | carrar | haha |
| 01:35.55 | drmessano | So Windows SBS + 10 cals is the price of a workstation |
| 01:35.58 | carrar | If you can' |
| 01:35.59 | carrar | err |
| 01:36.05 | drmessano | 10 additional.. 15 total |
| 01:36.07 | carrar | If you can't do it on UNIX you shouldn't be doing it |
| 01:36.29 | staffmember | freetown2: do you have any suggestions for me |
| 01:36.38 | freetown2 | <PROTECTED> |
| 01:36.47 | drmessano | If a business needs AD, they wouldnt shy at dropping that.. otherwise, they dont need and wouldnt implement a FOSS domain controller |
| 01:37.03 | carrar | staffmember, why not route calls over a PRI to your nortel? |
| 01:37.08 | carrar | from Asterisk |
| 01:37.20 | freetown2 | staffmember, well, you can write dialplans to hand calls over to nortel exts i believe...but you'd have to program the nortel too |
| 01:38.14 | drmessano | Just like "Office 2007 doesnt run well under Wine in Ubuntu" <-- YOURE DOING IT WRONG |
| 01:38.24 | staffmember | thats the problem, thats why i need some docs, i dont have a problem with asterisk, i feel comfortable with the sytem, but i have no nortel knowledge |
| 01:38.40 | freetown2 | drmessano, i'd implement samba any day over WIndows if it had group policy support. |
| 01:38.44 | staffmember | on the asterisk side im fine, i just need any whitepapers or docs regarding a nortel/asterisk setup |
| 01:39.24 | carrar | staffmember, hire someone for 2 hours to config the nortel to send a extension range to your asterisk and accept calls as well |
| 01:39.28 | drmessano | freetown2: You would be in the minority |
| 01:39.34 | freetown2 | staffmember, i doubt that you'd get manuals for nortel from the * site... |
| 01:40.00 | carrar | Or just stick Asterisk inbetween nortel and the outside |
| 01:40.18 | freetown2 | drmessano, yeah, how many Windows admins would move to Linux anyway... |
| 01:40.18 | staffmember | yeah, well thats what our endresult would be |
| 01:40.23 | staffmember | or atleast thats what we wwant as the end result |
| 01:40.29 | staffmember | have asterisk sit between nortel and the outside |
| 01:40.41 | carrar | so do that first |
| 01:40.43 | staffmember | we have 4 fxo/fxs on the asterisk side, we only have 3 lines going into the nortel |
| 01:40.52 | staffmember | okay |
| 01:40.57 | *** join/#asterisk neurosys (n=neurosys@c-71-196-20-208.hsd1.fl.comcast.net) |
| 01:41.01 | staffmember | let me try google and see if i can find any white papers on nortel/* |
| 01:41.25 | carrar | Just have asterisk pass everything to begin with |
| 01:41.34 | carrar | then peal out things as you need them |
| 01:41.36 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 01:42.42 | freetown2 | is depressed he did not get to implement * for the new school telephone system because HQ ordered a Cisco solution by Datacraft |
| 01:42.57 | staffmember | cisco solution can get prtty expensive |
| 01:43.14 | staffmember | i was going through the same thing, i just guaranteed that i can use cisco phones on asterisk |
| 01:43.19 | freetown2 | the school got preferential bulk pricing apparently |
| 01:43.20 | staffmember | pricing of * sold my boss instantly |
| 01:43.30 | staffmember | how many phones in the schoo l ? |
| 01:43.35 | freetown2 | 50 |
| 01:43.38 | p3nguin | it's hard to beat free, eh? |
| 01:44.12 | freetown2 | a man at the top said something about only freetown2 knowing how it would work.... |
| 01:44.13 | carrar | Phones aren't free |
| 01:44.18 | carrar | but cheap |
| 01:44.20 | p3nguin | Asterisk is. |
| 01:44.34 | Corydon76-dig | Asterisk isn't a phone. |
| 01:44.42 | carrar | but they will need phones |
| 01:44.55 | staffmember | close to impossible to beat free especially when you can get all the features you want |
| 01:44.55 | p3nguin | Y'all need to pay better attention. |
| 01:44.58 | carrar | 50 of them |
| 01:45.01 | carrar | at least |
| 01:45.32 | carrar | still massively cheaper then crisco |
| 01:45.36 | staffmember | yeah |
| 01:45.42 | staffmember | cisco phones can get expensive if your going with higher end stuff |
| 01:45.43 | p3nguin | You'll be buying phones regardless of which PBX you decide to use. |
| 01:45.54 | carrar | or $horeTel |
| 01:46.11 | carrar | Buy use cisco phones |
| 01:46.14 | *** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) |
| 01:46.44 | carrar | refurbs |
| 01:47.37 | freetown2 | we will be paying around 73USD for 7906G phones |
| 01:47.57 | p3nguin | That's a good buy if you meant 7960G. |
| 01:47.58 | mchou | than include the sip license? |
| 01:48.06 | freetown2 | 153USD for 7942G |
| 01:48.08 | carrar | no need |
| 01:48.08 | mchou | that* |
| 01:48.11 | carrar | use SCCP |
| 01:48.46 | freetown2 | 194USD for 7962G |
| 01:50.01 | *** join/#asterisk jakent (n=john@c-69-143-184-201.hsd1.va.comcast.net) |
| 01:50.24 | *** join/#asterisk LemensTS (n=customgt@adsl-70-238-130-71.dsl.stlsmo.sbcglobal.net) |
| 01:50.25 | freetown2 | but I cannot for the life of me understand why we have to pay for 130 callmanager device licenses. Ah well, HQ laid down the law. |
| 01:50.51 | freetown2 | goes for a drink |
| 01:50.54 | carrar | and you'll probably need to pay for conferencing seat lic |
| 01:50.57 | carrar | etc... |
| 01:51.00 | carrar | voicemail |
| 01:51.13 | p3nguin | Someone was here the other day that said you should buy "spare" phone so you don't have to pay for the licenses. |
| 01:51.30 | freetown2 | yeah, lots of stuff. all of which we could have got for free and with unlimited space |
| 01:52.00 | carrar | yup |
| 01:52.05 | p3nguin | But if you're going to be using CME or something, you'll be required to have the licenses, where with Asterisk you wouldn't. |
| 01:52.09 | carrar | fire your asterisk sales person |
| 01:52.12 | freetown2 | not versed at all in cisco products - i don't understand what that meant |
| 01:52.28 | freetown2 | carrar, uh...myself? no way! |
| 01:52.31 | carrar | heh |
| 01:52.34 | freetown2 | liek i said |
| 01:52.37 | freetown2 | a man at the top said something about only freetown2 knowing how it would work.... |
| 01:52.52 | freetown2 | HQ can be dumb sometimes |
| 01:53.00 | carrar | put instructions on a internal wiki |
| 01:53.09 | p3nguin | There's always someone else that knows how to manager Asterisk if freetown2 ever left. |
| 01:53.22 | p3nguin | For example, I would have done it. |
| 01:53.41 | freetown2 | i emphasized that there are quite a few companies in HK that can handle * |
| 01:54.15 | carrar | Cisco might be a bad choice in Asia |
| 01:54.24 | p3nguin | You don't even need an entire company, really. One or two admins could keep a 50-phone school going. |
| 01:54.25 | carrar | they are completely restructuring their company |
| 01:54.28 | carrar | in Asia |
| 01:54.40 | netpro25_ | Anyone familiar with IAXmodem? |
| 01:54.41 | freetown2 | right now, we have gone while past the original implementation date and Datacraft came up with some really funny stuff lately...maybe I might get to do * in the end :P |
| 01:54.45 | ManxPower-work | The North American Numbering Plan (NANP) will welcome a new member country in 2010. Joining the NANP will be Sint Maarten, the southern portion of the island of Saint Martin/St. Martin. Sint Maarten was formerly part of the Netherlands Antilles; its northern portion, called Saint Martin, is an overseas collectivity of France. |
| 01:54.50 | p3nguin | netpro25_: barely. Just ask your questions. |
| 01:55.08 | carrar | damn I need to update NPANXX |
| 01:55.13 | netpro25_ | Asterisk says it is registering but it is saying Registration failed. |
| 01:55.42 | freetown2 | carrar, you got a link? I'd love to inform the school management that maybe HQ does not know what is best afterall :P |
| 01:55.53 | p3nguin | netpro25_: Paste your configs into a pastebin so we can see what is wrong. |
| 01:56.05 | netpro25_ | k |
| 01:56.15 | freetown2 | p3nguin, ah...i am not sure how many * admins there are in HK |
| 01:56.32 | p3nguin | freetown2: Put the box online and I'll do it from the US. |
| 01:56.38 | freetown2 | and they'd have to be willing to do a lot more than just * if I left |
| 01:57.12 | freetown2 | p3nguin, sure...if you could convince the school to let you after i leave :P |
| 01:57.20 | carrar | freetown2: http://newsroom.cisco.com/dlls/2010/corp_011110.html |
| 01:57.26 | freetown2 | or if i suffer an accident... |
| 01:57.36 | carrar | Sounds like a shakeup |
| 01:57.48 | p3nguin | freetown2: I doubt they would like me -- I know English. |
| 01:57.56 | freetown2 | now if only it was on the scale of Nortel.... |
| 01:58.00 | carrar | Engirsh! |
| 01:58.06 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 01:58.12 | freetown2 | p3nguin, ESF school here. You'd be more than welcome |
| 01:58.18 | freetown2 | www.esf.edu.hk |
| 01:59.36 | carrar | redhat |
| 02:00.25 | carrar | You're 2 release behind in php |
| 02:00.29 | carrar | heh |
| 02:01.04 | carrar | ZendServer needs updating |
| 02:03.01 | *** join/#asterisk timholum (n=chatzill@209-206-141-151.dyn.centurytel.net) |
| 02:06.18 | p3nguin | Apache needs it, too. |
| 02:06.30 | carrar | yeah, whole system |
| 02:06.52 | p3nguin | Or just wipe it out and go with OpenBSD. |
| 02:07.51 | timholum | I am having issue's with connecting to my asterisk box through my nat, I am using x-lite for a sip phone i have nat and qualify enabled in my sip.conf, my phone registers and makes calls with sound if i am on the local network but when I have my sip phone on a different network like at home or something it registers allows me to make calls but sound does not work. any ideas |
| 02:07.58 | p3nguin | issue's? |
| 02:08.05 | p3nguin | Really? |
| 02:08.36 | timholum | sorry, i suck a writting :) and english in genral |
| 02:08.54 | *** join/#asterisk etfonhomey (n=etfonhom@173-8-16-185-WashingtonDC.hfc.comcastbusiness.net) |
| 02:09.49 | carrar | ~sipnat |
| 02:09.49 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 02:10.05 | p3nguin | Alright. It is possible that your routing appliances are breaking SIP on those remote locations. There are other things that could be at fault, but everything would be just a guess until you provide some configs and debug information. |
| 02:15.35 | p3nguin | freetown2: Is that your server? |
| 02:16.12 | *** join/#asterisk opticy (i=ycitpo@that.violates.us) |
| 02:17.12 | timholum | carrar: thanks, It was I forgot to port forward 10000 - 20000 to my asterisk box |
| 02:21.04 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 02:22.21 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 02:22.21 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 02:23.57 | freetown2 | sorry, had to handle a call. |
| 02:24.07 | freetown2 | www.esf.edu.hk = HQ's baby |
| 02:24.19 | *** join/#asterisk voipmonk (n=shido6@67.204.37.228) |
| 02:24.22 | p3nguin | Not under your control, then? |
| 02:24.28 | freetown2 | definitely NOT |
| 02:24.31 | freetown2 | :D |
| 02:24.51 | p3nguin | You need to let them know that you found some outdated and vulnerable software during a routing security audit. |
| 02:25.01 | freetown2 | we got a new moodle server running Centos 5.4 - I had to tell the HQ sysadmin to update the thing |
| 02:25.05 | p3nguin | s/routing/routine/ |
| 02:25.43 | freetown2 | ponders over his job security at ESF |
| 02:25.50 | freetown2 | nah |
| 02:26.08 | ManxPower-work | ~esf |
| 02:26.45 | ManxPower-work | ~esf |
| 02:26.46 | infobot | esf is, like, Extended Super Frame, the most common framing method used for T-1s connected to Asterisk. |
| 02:27.31 | freetown2 | i have been vocal enough about the abysmal mail system a HQ chosen vendor implemented. Not sure I want to start with members of the HQ IT team... |
| 02:27.59 | freetown2 | after all, my job is at Bradbury School - not HQ |
| 02:28.19 | p3nguin | oh |
| 02:28.27 | p3nguin | Well, screw 'em, then. |
| 02:29.33 | *** join/#asterisk darkskiez__ (n=dz@62-50-207-156.client.stsn.net) |
| 02:29.39 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 02:29.47 | p3nguin | applies for a job with HQ |
| 02:29.49 | freetown2 | work culture here in HK could do with a lot of improvement |
| 02:29.50 | *** join/#asterisk pawz (n=pawz@210.56.87.151) |
| 02:30.23 | freetown2 | hopes p3nguin will remember him if he does get into HQ to make a big shakeup :D |
| 02:30.58 | p3nguin | compiles apache |
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| 03:15.32 | phoneLine | hey all, anyone know ssmtp? |
| 03:18.42 | phoneLine | I cant get my ssmtp to work with gmail... it appears to not even attempt starttls |
| 03:19.36 | Nugget | yeah, that's just gmail being gmail. |
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| 03:45.10 | phoneLine | think I found it out... my host allows some crazy redirect, so its a three step process but Im getting my emails |
| 04:05.59 | phoneLine | Anyone know how I can change the from name?? It appears to come from "Asterisk PBX" |
| 04:09.31 | *** join/#asterisk DarkFibre (n=dmelouk@127.159.119.70.cfl.res.rr.com) |
| 04:14.19 | p3nguin | Talking about voicemail emails? |
| 04:15.16 | p3nguin | If yes, voicemail.conf - serveremail |
| 04:15.41 | p3nguin | and voicemail.conf - fromstring |
| 04:15.57 | p3nguin | phoneline: ^ |
| 04:19.12 | *** join/#asterisk OrNix (n=ornix@l151-249-47.static.cn.ru) |
| 04:26.43 | phoneLine | fanx p3nguin! |
| 04:30.13 | p3nguin | http://www.pidjin.net/2009/12/16/wax-up/ |
| 04:33.58 | *** join/#asterisk chendy_ (n=chatzill@119.139.169.21) |
| 04:38.50 | *** join/#asterisk runic (i=4b97cff9@gateway/web/freenode/x-slhiwfljohucujqa) |
| 04:39.07 | runic | heya |
| 04:39.37 | p3nguin | moo |
| 04:39.54 | LemensTS | meep |
| 04:40.00 | freetown2 | baa |
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| 04:43.22 | runic | anyone pro with * behind cisco IOS fw/nat ? |
| 04:43.34 | runic | having no audio issues from outside the nat |
| 04:43.41 | runic | "no audio" rather |
| 04:44.07 | freetown2 | * and nat again. |
| 04:44.45 | p3nguin | You'll see it a lot. |
| 04:44.49 | p3nguin | ~sipnat |
| 04:44.50 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:46.20 | p3nguin | Make sure your ports are forwarded properly. |
| 04:47.08 | |Rain| | runic: I had an issue a while back where the router was being too helpful and breaking asterisk's nat work arounds -- I enabled nat=yes in asterisk and disabled the router mangling with 'no ip nat service sip udp port 5060' to fix it |
| 04:48.05 | runic | right now I have no ip nat service sip udp port 5060, and nat=yes on ast |
| 04:48.38 | p3nguin | Check your RTP ports. |
| 04:48.38 | runic | doing the nat pool trick for the 10000+ as well (limited it to 10000 - 10050 in the rtp.conf, assuming that was ok) |
| 04:49.06 | p3nguin | Make sure you match your rtp.conf ports to what you have forwarded. |
| 04:49.13 | runic | yeah, done that |
| 04:49.19 | p3nguin | oh, misread |
| 04:49.21 | runic | i'll double check everything again though |
| 04:50.55 | p3nguin | RTP uses 2 ports per call, so that'll limit you to 25 calls at once. I guess that's probably enough for you. |
| 04:52.37 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 04:53.17 | runic | yeah its for the home, more than enough |
| 04:53.24 | runic | thanks for the tip though, didn't know that |
| 04:53.41 | runic | i've got alot to learn still, just getting started (well, casually, over the past year) |
| 04:54.30 | runic | originally, magicjack hax got me interested |
| 04:57.41 | *** join/#asterisk jetlag (i=jetlag@pool-70-104-75-14.pskn.east.verizon.net) |
| 05:05.38 | runic | whups |
| 05:05.41 | runic | where'd nat=yes go |
| 05:05.52 | runic | gonna give it another go 8+) |
| 05:06.46 | freetown2 | in sip.conf in an appropriate section |
| 05:09.18 | runic | Lately i've been rolling with PBXIAF, i guess it would go in sip_general_custom.conf, but don't hate on me (i don't know how frowned upon distributions are, i'm all asterisk nub) |
| 05:09.51 | runic | I did build asterisk from scratch and got it working with magicjack's service awhile ago, but wanted to test something more turnkey |
| 05:15.36 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
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| 05:18.40 | runic | oks o |
| 05:18.42 | runic | err* |
| 05:18.46 | runic | if I do a sip show peers |
| 05:19.05 | runic | i'm seeing my external sip client registered on a port that is not in my rtp.conf |
| 05:19.17 | runic | does rtp.conf not dictate where the client negotiates the port? |
| 05:19.35 | p3nguin | ~freepbx |
| 05:19.36 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 05:20.06 | runic | i dont think this is a freepbx issue, i'm careful not to touch things that are "managed" by freepbx |
| 05:20.13 | runic | i'm working from shell atm |
| 05:20.50 | drmessano | rtp.conf is the audio |
| 05:20.58 | drmessano | rtp is the audio, i should say |
| 05:21.28 | p3nguin | I was just mentioning it as a response to your earlier statement about PiaF. |
| 05:21.43 | runic | which is my problem, and in "sip show peers" its showing my client connected on port 10224 during an active call |
| 05:22.10 | p3nguin | Did you remember to reload the things you have changed? |
| 05:22.11 | runic | i guess i'm just not sure if "Port" in sip show peers is the audio or not |
| 05:22.13 | runic | yeah |
| 05:22.17 | runic | absolutely |
| 05:22.19 | drmessano | Its not |
| 05:22.19 | *** join/#asterisk roo (n=w7@freenode/staff/wikimedia.werdan7) |
| 05:22.23 | runic | ok |
| 05:22.37 | drmessano | NAt stuff thats too hard to explain in 140 characters or less |
| 05:22.46 | freetown2 | would anybody here dream of using 50 wireless sip phones in a small campus that has 48 access points blanketing it? |
| 05:23.25 | freetown2 | wireless-n network if anybody asks |
| 05:23.30 | p3nguin | Not likely. Most people do not like wifi and sip in the same sentence unless it is to say not to use them together. |
| 05:23.55 | freetown2 | phones wont be moving around |
| 05:24.03 | drmessano | So? |
| 05:24.05 | p3nguin | Then wire them. |
| 05:24.15 | p3nguin | Wireless is for mobility. |
| 05:24.16 | freetown2 | laptops will be on 5ghz radio band and phones on the 2.4ghz radio band |
| 05:24.18 | drmessano | Its still SIP and still WIFI, and SIP wifi sucks |
| 05:26.04 | runic | it looks like my sip client keeps connecting to port 10224 |
| 05:26.09 | runic | in sip show peers |
| 05:26.20 | runic | i've even tried multiple sip clients (albeit from same IP address) |
| 05:26.43 | p3nguin | to it or from it? |
| 05:27.07 | runic | to the * box |
| 05:27.07 | p3nguin | Check your port= in sip.conf, maybe. |
| 05:27.48 | runic | 5060 for the extension |
| 05:28.21 | runic | my point is, if it were autonegotiated or something, is there a way to make asterisk "forget" ? like database delete or something? or is that the wrong path to take |
| 05:28.45 | p3nguin | for the extension... |
| 05:28.48 | runic | yeah |
| 05:28.51 | p3nguin | extensions do not have ports. |
| 05:29.22 | runic | i mean, its specified under the extension in the sip.conf |
| 05:29.28 | p3nguin | Extensions are found in extensions.conf |
| 05:29.53 | p3nguin | You're probably talking about a device. |
| 05:29.58 | runic | sorry, I mean the sip device |
| 05:29.59 | p3nguin | Devices are found in sip.conf. |
| 05:30.07 | runic | thanks for clarification |
| 05:30.25 | p3nguin | That'll save you from the embarassment later. :) |
| 05:31.08 | phoneLine | the port is auto negotiated... similar to an http browser... the initial com is on 5060 tho |
| 05:32.16 | p3nguin | Odd. All of mine say the port that is configured in the device. |
| 05:32.46 | phoneLine | and as for that many sip clients, Id never do it... I would use the AASTRA wireless system |
| 05:32.52 | p3nguin | usually 5060, but some on other ports... but all on whatever port is configured in the device. |
| 05:33.34 | phoneLine | yes |
| 05:34.00 | phoneLine | and the aastra device Im talking about it hte MBU 400, I saw it in R&D / pre release and it was sick! |
| 05:34.17 | runic | i have no trouble with phones internally |
| 05:34.31 | runic | my goal is to get my iphone sip client working remotely, which is where the trouble is |
| 05:34.56 | runic | but switching clients and what not, seems odd they'd negotiate at that port |
| 05:35.09 | runic | same port |
| 05:37.01 | runic | looking at my router thats the client's port, actually |
| 05:39.17 | titter | runic: are you trying to get remote clients to nat properely? |
| 05:39.37 | runic | yessir |
| 05:41.17 | titter | runic: udp 5060 upd 10000-20000 if you are using iax udp 4569 |
| 05:42.40 | titter | runic: I would suggest testing remote connections with something such as X-lite to make sure they are working. Most mobile phone clients can be iffy at best. I have had success with my Android phone's SIP client |
| 05:43.53 | *** join/#asterisk themolester (i=themoles@cpe-173-171-165-200.tampabay.res.rr.com) |
| 05:44.34 | runic | fair enough, thanks |
| 05:44.41 | themolester | what would cause asterisk to hang up a call after 1s on outgoing calls only (sip2sip) |
| 05:45.13 | themolester | incoming calls work fine, and the call does go through, if you talk during the first second you can hear eachother |
| 05:46.59 | titter | themolester: can you pastebin the console |
| 05:48.18 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 05:48.46 | themolester | give me a minute, need to take out the sensitive info |
| 05:49.12 | ChannelZ | that's OK, we like pictures of boobies |
| 05:49.38 | *** join/#asterisk dudee (n=dude@119.75.10.98) |
| 05:54.50 | dudee | Hi, i am having some problems, when i dial in internal route, it takes almost 5 solid seconds before it rings the extension, is there some timeout issue it's waiting for? its a DAHDI interface. any recommendations ? |
| 05:57.54 | [TK]D-Fender | dudee: Yes, if then number you are dialing COULD be part of something longer that is also legal then it will wait |
| 05:58.19 | [TK]D-Fender | dudee: So make sure that you don't have unwanted over-generic patterns in your dialplan. |
| 05:59.34 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-mccvkxathazdvdfs) |
| 06:01.06 | titter | [TK]D-Fender: have any clue why I need to unplug my fxo port on my analog card, and plug it back in after restarting my server? |
| 06:01.42 | [TK]D-Fender | titter: No |
| 06:02.43 | titter | it's weird it's a tdm410p card ... dahdi_tool shows it having 0 act chans ... as soon as I unplug it, and plug it back in ... it goes to 1 active |
| 06:03.36 | themolester | http://pastebin.ca/1750661 |
| 06:03.45 | *** join/#asterisk DarkFibre (n=dmelouk@127.159.119.70.cfl.res.rr.com) |
| 06:03.55 | themolester | titter ^^ |
| 06:04.28 | *** join/#asterisk pawz (n=pawz@210.56.86.102) |
| 06:05.25 | themolester | i did like 6 v |
| 06:05.56 | dudee | [TK]D-Fender: where do i find local dial plans? e.g. internal transfer |
| 06:06.30 | themolester | dudee i believe those are on the endpoint |
| 06:06.59 | themolester | usually http://endpointip/ or in your remote config |
| 06:07.10 | *** join/#asterisk chendy (n=chatzill@204.152.211.137) |
| 06:07.32 | titter | have you tried canreinvite=no |
| 06:07.57 | dudee | themolester: that i know |
| 06:08.01 | themolester | titter on the endpoint, or the trunk? |
| 06:08.22 | [TK]D-Fender | dudee: #freePBX <- Not supported here |
| 06:08.32 | titter | would be in the sip.conf where the phones are registering |
| 06:10.16 | themolester | i meant the section ... i believe they are were set under the extensions during that log, and I have also tried it under trunk |
| 06:12.35 | themolester | oh, if it helps, the ring tone during dial seemed... a bit off |
| 06:12.38 | titter | i have always set this in the context with the extension |
| 06:12.49 | themolester | almost like there were two ringers |
| 06:13.16 | themolester | but maybe just a bad connection, my testing was done with the cordless |
| 06:14.49 | themolester | titter in that case, it's set |
| 06:15.25 | themolester | i only have a basic understanding of the communication going on, does it seem like it is reinviting anyway? |
| 06:16.39 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 06:17.12 | *** part/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 06:20.08 | themolester | titter any other ideas? |
| 06:20.13 | *** join/#asterisk Natlonehat (n=Enrage@202.170.42.67) |
| 06:20.38 | Natlonehat | trying to set up func_odbc |
| 06:21.00 | Natlonehat | database query gives errors. |
| 06:22.20 | Corydon76-dig | Natlonehat: can you use: isql dbname |
| 06:22.36 | Corydon76-dig | where dbname is a configuration in /etc/odbc.ini ? |
| 06:22.36 | Natlonehat | yup |
| 06:22.50 | Natlonehat | it's the query I'm having trouble with Croydon |
| 06:23.07 | Corydon76-dig | Does your func_odbc dsn point to an entry in res_odbc.conf? |
| 06:23.23 | Natlonehat | yup |
| 06:23.29 | Natlonehat | here's the actual issue: |
| 06:23.31 | Natlonehat | read=SELECT s.ServerName FROM asteriskrouter.Stats s inner join (asteriskrouter.Dnis d) on (d.Programs=s.Program) WHERE d.Dnis='${SQL_ESC(${ARG1})}' |
| 06:23.31 | Natlonehat | <PROTECTED> |
| 06:24.18 | Natlonehat | the mysql query works from the mysql gui, but when i try to plug this into a read = statement |
| 06:24.34 | Natlonehat | and then call it from dialplan, |
| 06:24.46 | Natlonehat | cant get it to work |
| 06:24.55 | Corydon76-dig | What's the error? |
| 06:25.45 | Natlonehat | it's asking for an '=' because the statement doesnt fit in 1 line maybe? |
| 06:26.11 | VxJasonxV | I have an odd question. Say I have a physical phone at work, but want a phone at home, whether soft phone or cell phone app is irrelevant. I don't have access to my phone at work, and I'm not going to remember to unplug it every day. |
| 06:26.29 | VxJasonxV | I already know from testing that two devices will have a registration war if logging in as the same extension/user. |
| 06:26.40 | Corydon76-dig | Natlonehat: what version are you using? |
| 06:26.48 | Natlonehat | ast 1.4.24 |
| 06:26.53 | Natlonehat | mysql 4.1 |
| 06:26.58 | VxJasonxV | So, without doing something like followme and specifying an external phone number, is there a way to register multiple devices to the same extension sanely? |
| 06:27.10 | Natlonehat | @ VxJasonxV I do this all the time |
| 06:27.14 | Natlonehat | the thing is |
| 06:27.34 | Natlonehat | your desk phone wont get calls while you've got the soft phone registered |
| 06:27.42 | VxJasonxV | that's fin |
| 06:27.43 | VxJasonxV | +e |
| 06:28.00 | Natlonehat | and then you've got to force it to re-register [if you dont have an auto-timeout option on the phone] |
| 06:28.06 | VxJasonxV | that's the thing |
| 06:28.10 | VxJasonxV | the phone does, 30 minutes |
| 06:28.11 | Corydon76-dig | VxJasonxV: Sure, as long as they're registering to two separate devices |
| 06:28.19 | VxJasonxV | hence the "registration war" |
| 06:28.42 | Natlonehat | well...you could try 'hot desking' |
| 06:28.42 | Corydon76-dig | In your extension, you do a Dial(SIP/myphoneatwork&SIP/myphoneathome) so that it rings in both places |
| 06:28.49 | VxJasonxV | Corydon76-dig, pardon my ignorance, but could you elaborate on "registering to two seperate devices"? Do you mean two different asterisk servers/endpoints entirely? |
| 06:29.01 | VxJasonxV | I was hoping to only have one extension period. |
| 06:29.06 | Corydon76-dig | Same server, different SIP peers |
| 06:29.14 | VxJasonxV | can two peers have the same extension? |
| 06:29.19 | Corydon76-dig | VxJasonxV: You DO have only one extension |
| 06:29.28 | Corydon76-dig | See the Dial line above |
| 06:29.39 | VxJasonxV | yeah, I need to work on my terminology |
| 06:29.47 | Corydon76-dig | Two devices, one extension |
| 06:29.47 | VxJasonxV | I was hoping to have the same ... peer address? |
| 06:29.52 | Corydon76-dig | Nope |
| 06:30.00 | Corydon76-dig | Not going to happen |
| 06:30.05 | VxJasonxV | how unfortunate |
| 06:30.15 | VxJasonxV | especially nowadays where you can have sip everywhere |
| 06:30.18 | Natlonehat | what phone is it? |
| 06:30.29 | Natlonehat | the hardphone i mean |
| 06:30.35 | VxJasonxV | Polycom 321 |
| 06:30.51 | VxJasonxV | I know I can increase the registration timeout, but bleh |
| 06:30.58 | Corydon76-dig | The SIP people who came up with that idea were smoking crack at the time. It's extremely impractical in the real world to have multiple devices register to the same device |
| 06:31.01 | VxJasonxV | we already have a base privisioning configuration |
| 06:31.18 | VxJasonxV | I've never smoked crack :/ |
| 06:31.21 | VxJasonxV | and why is it impractical? |
| 06:31.33 | Natlonehat | ok....not much experience with Polycom, but there's a neat feature in THomson phones that allow time based registrations. :) |
| 06:31.42 | Corydon76-dig | If you had a perfect network that never dropped packets and perfect phones that never spontaneously rebooted |
| 06:31.54 | Natlonehat | which you can use to restrict users from using the phone at times |
| 06:31.55 | Corydon76-dig | It might work fine. Unfortunately, the real world has problems |
| 06:31.59 | Natlonehat | that might come in handy |
| 06:32.15 | Natlonehat | @ Croydon76: agreed |
| 06:32.31 | VxJasonxV | I don't understand what phones never rebooting and dropped packets has to do with it. If the server would just remember registrations, the first one that answers can handle the media path invitation madness and what not |
| 06:32.48 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 06:33.02 | Natlonehat | you could do that with a bit of dialplan & database. |
| 06:33.15 | Natlonehat | just need to think it out properly |
| 06:33.29 | Corydon76-dig | VxJasonxV: because if the phone doesn't drop its registration, each time it goes offline for whatever reason, when it comes back up, it will give yet another registration |
| 06:33.31 | Natlonehat | oh & realtime |
| 06:33.41 | Natlonehat | with cache on |
| 06:33.42 | drmessano | I rather have each device as a seperate entity anyway |
| 06:33.46 | Corydon76-dig | In no time at all, you'll have a SIP peer that has 2 million registrations |
| 06:33.56 | VxJasonxV | (1) sane timeout limit |
| 06:34.01 | Corydon76-dig | You think that might consume resources? |
| 06:34.21 | VxJasonxV | if a phone is configured to register every 1800 seconds, the server should drop it shortly after that if there hasn't been a registration |
| 06:35.02 | Corydon76-dig | It also means that the server will send duplicate INVITEs, each of which will have unique tags |
| 06:35.21 | Corydon76-dig | Want the phone to get calls in triplicate? |
| 06:36.17 | Corydon76-dig | It's just a really dumb idea, and it's not going to be implemented here. Use multiple devices and tie them together at the extension. It's good stuff, and it works well. |
| 06:36.41 | Natlonehat | theres a thought. In short, why not set up a seperate SIP peer & get your extension to go to that at specified timees [execiftime] |
| 06:36.59 | Natlonehat | that way, users only know 1 extension that they have to dial |
| 06:37.03 | titter | Corydon76-dig: you familar with the TDM410P analog cards? |
| 06:37.14 | Corydon76-dig | vaguely |
| 06:38.30 | Natlonehat | If i might jump in for a sec....the database query for func_odbc...does that have to be just one line? |
| 06:38.37 | Corydon76-dig | I'm using an RMAed NFR TDM400P here, if that helps |
| 06:39.04 | titter | basically whenever my server reboots, I have to unplug my fxo port, and plug it back in ... dahdi_tool shows 0 active channels, then will go to 1 after the unplug and plug back in |
| 06:39.21 | titter | i got the card from the asterisk advanced course at digium |
| 06:39.28 | titter | figured i'd test it at home |
| 06:39.58 | Corydon76-dig | Sorry, no idea |
| 06:41.46 | Natlonehat | can someone help me with the db query: [PROGNAME] |
| 06:41.46 | Natlonehat | dsn=asterisk |
| 06:41.46 | Natlonehat | read=SELECT s.ServerName FROM asteriskrouter.Stats s inner join (asteriskrouter.Dnis d) on (d.Programs=s.Program) WHERE d.Dnis='${SQL_ESC(${ARG1})}' |
| 06:41.46 | Natlonehat | <PROTECTED> |
| 06:42.01 | Natlonehat | func_odbc |
| 06:42.12 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 06:43.37 | Corydon76-dig | Natlonehat: It should output a WARNING specifying the query it tried to execute. Could you pastebin that WARNING? |
| 06:43.59 | Natlonehat | ] WARNING[23028]: config.c:806 process_text_line: No '=' (equal sign) in line 45 of |
| 06:44.49 | Corydon76-dig | Natlonehat: oh, what you probably did is that there's a newline in the middle of the query. It has to all be on one line |
| 06:45.33 | Natlonehat | so if the query is longer than one line...? |
| 06:45.44 | Natlonehat | how do i deal with that? |
| 06:46.01 | Corydon76-dig | It'll wrap in the editor, but as long as it's all on one line in the editor, it'll work |
| 06:46.23 | Corydon76-dig | Oh, and no more than 2047 bytes long |
| 06:46.32 | Natlonehat | ok. |
| 06:46.36 | Natlonehat | let me try that. |
| 06:46.39 | Natlonehat | brb |
| 06:46.58 | themolester | Corydon76-dig if he wanted to make it two lines, is he able to escape the newline with \ or anything |
| 06:47.10 | themolester | I'm only half paying attn, reading my log |
| 06:47.12 | Corydon76-dig | themolester: not in anything short of 1.6.2 |
| 06:47.25 | *** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id) |
| 06:47.32 | Corydon76-dig | In 1.6.2, config files gain += for appending to a previous option |
| 06:48.14 | themolester | I'll try to remember that for future reference :) |
| 06:48.25 | themolester | Corydon76-dig any idea on my question from earlier? |
| 06:48.54 | Natlonehat | [Jan 14 17:48:04] WARNING[5388]: res_odbc.c:105 ast_odbc_prepare_and_execute: SQL Execute returned an error -1: 42000: [unixODBC][TCX][MyODBC]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'read=SELECT s.ServerName FROM asteriskrouter.Stats s inner join (222) |
| 06:49.25 | Corydon76-dig | Natlonehat: Pastebin your entire file |
| 06:49.27 | Corydon76-dig | ~pb |
| 06:49.28 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 06:49.47 | Natlonehat | ok. |
| 06:50.08 | Corydon76-dig | themolester: you assume that I've seen your question |
| 06:50.31 | themolester | Sorry for the assumption :) http://pastebin.ca/1750661 |
| 06:50.48 | themolester | I'm having an issue on outgoing calls only, using a sip trunk |
| 06:51.00 | themolester | call works (have audio) but drops after 1s |
| 06:51.14 | themolester | incoming works, can use it for hours |
| 06:51.47 | themolester | very long logfile... asterisk can get very talkitive in <2seconds |
| 06:52.15 | Natlonehat | Corydon: http://pastebin.com/m7e9605af |
| 06:52.47 | themolester | Natlonehat you have read= twice |
| 06:53.16 | themolester | you are passing "read=" as part of the sql query |
| 06:56.22 | Corydon76-dig | themolester: Dunno, sorry |
| 06:57.03 | Natlonehat | @ Themo....Thanks!! ***ecstatic**** call it pressure...been in the office 12 hours straight. |
| 06:57.05 | Natlonehat | :) |
| 06:58.00 | themolester | i'm really not sure what I'm reading in the log, but what it looks like to me is: asterisk gets confirmation that the call is connected, then immediately sends a bye (disconnect) |
| 06:58.09 | themolester | but i don't see anywhere that explains why |
| 06:58.20 | themolester | that about what you see? |
| 07:00.10 | themolester | Natlonehat no prob, glad to help where I can, I know how you feel... work sucks |
| 07:00.55 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 07:01.05 | themolester | I go to work, mess around with asterisk, banging my head against the desk, staring at the clock... come home, and excitedly set up my new voip dect phone |
| 07:01.07 | themolester | :) |
| 07:03.48 | Natlonehat | @ themolester & Croydon: quick question...so how many connections max can i set up simultaneously to mysql via func_odbc before i run into trouble? |
| 07:04.08 | Corydon76-dig | On default configuration, 100 |
| 07:04.22 | Corydon76-dig | but you can up the limit in /etc/my.cnf |
| 07:04.28 | Natlonehat | ok. |
| 07:04.40 | Natlonehat | do i need to dfine pooling in res_odbc? |
| 07:05.08 | Corydon76-dig | Yes, you probably want pooling for MySQL |
| 07:05.13 | Natlonehat | & how do i up the limit in my.cnf |
| 07:05.43 | Corydon76-dig | It's not truly necessary, but I've found that using 1 query per connection generally makes the mysql libraries more stable |
| 07:05.56 | Natlonehat | ok. |
| 07:06.04 | Corydon76-dig | You'll need to consult with the mysql documentation for that |
| 07:06.22 | Natlonehat | really appreciate your assistance corydon. |
| 07:06.24 | Natlonehat | :) |
| 07:06.59 | Natlonehat | [asterisk] |
| 07:06.59 | Natlonehat | enabled => yes |
| 07:06.59 | Natlonehat | dsn => asterisk-connector |
| 07:06.59 | Natlonehat | pre-connect => yes |
| 07:06.59 | Natlonehat | pooling => yes |
| 07:06.59 | Natlonehat | limit => 100 |
| 07:07.05 | Natlonehat | does this look ok? |
| 07:07.24 | Corydon76-dig | Yes, but in the future, please use pastebin for something that large |
| 07:08.05 | Natlonehat | yes sir. [a bit new to irc] |
| 07:08.08 | Natlonehat | :) |
| 07:08.55 | Corydon76-dig | http://dev.mysql.com/doc/refman/5.1/en/server-parameters.html |
| 07:09.18 | Corydon76-dig | max_connections is the parameter you're looking for |
| 07:09.56 | Natlonehat | :) |
| 07:10.01 | Natlonehat | I'll go have a read now. |
| 07:22.56 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
| 07:24.42 | dudee | need some assistance. i have an issue. i dont think its a dial plan. both analog phones hooked up to asterisk using tdm2400p. problem is when trying to dial internal extension to internal extension, it takes 5 seconds before attempting to ring the extension. i dont have this issue for external to internal or from internal to external. only internal extension to internal extension. e.g. extension 551 calls extension 552, it takes 5 seconds before it even rings. |
| 07:27.27 | *** join/#asterisk creativx (n=creadure@197.82-134-19.bkkb.no) |
| 07:31.55 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 07:32.47 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 07:34.20 | ChannelZ | It might be your dialplan if you have any patterns for extensions, * might be waiting for additional digits in order to determine what the correct extension is |
| 07:34.42 | *** join/#asterisk Ad-Hoc (n=nimbus@62.169.216.185) |
| 07:34.45 | donatas | guys, can anyone show me how to configure IVR meniu ? |
| 07:35.14 | ChannelZ | For instance if you have an extension 55 and 555, if you dial 55 it will wait to make sure you aren't dialing 555. |
| 07:35.19 | *** join/#asterisk dpryo (n=hn@donatello.ondskap.net) |
| 07:35.57 | ChannelZ | donatas: read: |
| 07:36.00 | ChannelZ | ~book |
| 07:36.01 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 07:36.22 | dpryo | Hi. When asterisk parses DTMF (for either transfers or "stop recording" etc), some of the DTMF tone is sent through, just a few ms, but enough for the other end to notice.. Is there any way to adjust this? |
| 07:36.56 | dpryo | (inband dtmf) |
| 07:37.31 | *** join/#asterisk chendy_ (n=chatzill@119.139.170.244) |
| 07:43.50 | ChannelZ | dpryo: I don't think so |
| 07:44.43 | benngard | i am lost, dont know where to start looking, have a scenario like this: |
| 07:45.49 | benngard | ext 914 - avaya cm - ooh323 trunk asterisk - ext 0317998975 (sip phone) and 0317998989 (queue) |
| 07:47.44 | benngard | 914 calls 0317998975, 914 sees sip #588 in its display, makes sense, the name of the trunk is "sip" and trunk number is "588" in avaya cm |
| 07:48.37 | benngard | but when when 914 calls the queue 0317998989 it sees "Asterisk PBX" in its display, in some way * manages to get information back, but i cant figure out how |
| 07:50.49 | *** join/#asterisk sp3 (n=sp3@75-144-110-147-Indianapolis.hfc.comcastbusiness.net) |
| 07:51.04 | ChannelZ | you can set the CallerID (I think that's what you're asking) for any given channel |
| 07:51.32 | ChannelZ | like in sip.conf, callerid=Some Guy <914> |
| 07:52.10 | benngard | have a line like: callerid=Magnus Benngard <0317998975> at my extension |
| 07:53.03 | ChannelZ | (I should say 'any given device') - yes that should get passed on through any channels that support the concept of CallerID. I'm not sure specifically in the case of ooh323 |
| 07:54.43 | benngard | neither am i, but i found: in ooh323.conf, ;H323-ID to be used for asterisk server ;Default - Asterisk PBX, h323id=ObjSysAsterisk |
| 07:55.27 | benngard | so i wonder if i can change "ObjSysAsterisk" to something else |
| 07:57.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:01.53 | ChannelZ | shrugs - dunno anything about h323, if that ID might have some other significance |
| 08:05.31 | *** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au) |
| 08:11.26 | *** join/#asterisk lftsy (n=mlr@install.deckpoint.ch) |
| 08:12.57 | *** join/#asterisk donnib (n=aaa@0x555281d0.adsl.cybercity.dk) |
| 08:13.00 | donnib | hi |
| 08:13.29 | donnib | can one of you clever guys tell me what is OpenSER and what's the difference between that and asterisk ? |
| 08:13.42 | *** join/#asterisk jkroon (n=jkroon@dsl-244-25-144.telkomadsl.co.za) |
| 08:15.07 | jkroon | Can somebody please take a look at http://pastebin.co.za/76928, it's a SIP trace that I cannot see a problem with, yet the call was definitely "dropped". The reason given by eyeBeam is "User Hung Up" but this just doesn't seem correct. |
| 08:15.18 | tzafrir_laptop | donnib, for starters, OpenSER has been both renamed and forked, so now you have Kamailio and OpenSIPS |
| 08:15.42 | jkroon | I can't see anything wrong with that, thus can't imagine why eyeBeam would hangup the call. |
| 08:15.50 | donnib | tzafrir_laptop: ok so what are the difference between those and asterisk |
| 08:16.05 | tzafrir_laptop | Those are SIP proxies |
| 08:16.05 | jkroon | this didn't happen with asterisk 1.6.1.6 but has been confirmed to happen both with 1.6.1.11 and 1.6.1.12. |
| 08:16.06 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 08:16.13 | jkroon | donnib, sip server != sip proxy. |
| 08:16.40 | donnib | but what is the bennefit of using a sip proxy then ? |
| 08:16.50 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 08:17.06 | donnib | would a setup be sip client -> sip proxy -> asterisk ? |
| 08:18.36 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 08:18.58 | jkroon | donnib, yes. a proxy does something different, it takes a SIP address (user@domain) and figures out to which server to send it. |
| 08:19.34 | donnib | so all the extension stuff still relies in asterisk i mean the authentication of a sip client |
| 08:19.36 | donnib | ? |
| 08:19.39 | jkroon | a server does more complex things, like dialplan etc. the proxy just shuffles things along (very much like a squid proxy compared to apache) - you don't point your browser directly at a server and get all addresses from that server hey? |
| 08:19.50 | jkroon | that depends on your configuration. |
| 08:20.25 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:20.36 | donnib | if that's the case then i guess asterisk must communicate both ways with the proxy: asterisk <-> proxy |
| 08:21.01 | jkroon | again, it depends. |
| 08:21.31 | donnib | hmm ok, i am just trying to see the benefit of having a sip proxy and when it's needed, maybe load balancing ? |
| 08:21.41 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 08:21.51 | jkroon | i use just asterisk because i don't need the additional functionality of a proxy currently. |
| 08:22.11 | jkroon | that would be a legit use. |
| 08:22.24 | donnib | well yeah but inorder to know if i need the functionality i need to know what it can offer ;) |
| 08:22.28 | jkroon | well, you can't pass something like jkroon@uls.co.za to any asterisk server and expect the call to complete :) |
| 08:22.55 | donnib | no unless it's setup like that or it accepts anonymous calls |
| 08:23.27 | jkroon | i can promise you, asterisk will not complete that call. |
| 08:23.44 | jkroon | you can set it up any way you like, you're going to have trouble getting it to complete that call. |
| 08:24.09 | donnib | if you have an inbound route accepting jkroon as DID then what's the problem ? |
| 08:24.10 | jkroon | whereas a proxy would do the required SRV lookups, locate voip.uls.co.za and pass the call there. |
| 08:24.20 | jkroon | the @uls.co.za :) |
| 08:24.31 | donnib | ohh |
| 08:24.39 | donnib | i get it |
| 08:24.59 | donnib | yeah you probably right, so you can route calls based on address for instance |
| 08:27.11 | jkroon | yes, the idea with the proxy is that it'll take the complete address, eg jkroon@uls.co.za then look at the domain part (similar to an email gateway) and route the call based on the domain, the recipient server will then take care of the jkroon part. |
| 08:27.38 | donnib | ok now i know a little bit more, thx for clarifying that |
| 08:29.09 | jkroon | please note that my understanding and knowledge is also relatively rudementary, so I've explained it as I understand it. other people in this # may well point out a LOT of misunderstandings in the above. |
| 08:29.26 | donnib | sure, np i get the idea :) |
| 08:31.48 | jkroon | now if anybody can please take a look at that sip trace and let me know if they spot ANYTHING out of the ordinary. |
| 08:33.50 | donatas | how can i initiate external phone line ? i want to try ivr menu |
| 08:35.44 | ChannelZ | eh? |
| 08:35.52 | jkroon | my reaction exactly. |
| 08:36.20 | jkroon | donatas, you can "originate" a call into the applicable context perhaps? |
| 08:36.53 | jkroon | eg, originate SIP/your_phone_peer_name extension ???@??? |
| 08:37.51 | ChannelZ | I usually just make a dummy extension that does a Goto() to wherever I want for testing - like the 's' extension which is maybe what he's asking? |
| 08:39.50 | jkroon | he wants to call into his IVR from his "internal" phones. |
| 08:40.08 | jkroon | so yes, a dummy ext that Goto()s wherever an external call would have gone should work. |
| 08:40.45 | tzafrir_laptop | Which should do just that. Unless his dialplan does any fancy detection of the channel name |
| 08:41.34 | jkroon | if he was doing fancy crap like that he wouldn't be asking the question. |
| 08:42.46 | *** join/#asterisk dkirker-openmobl (n=dkirker@66-215-55-52.dhcp.snlo.ca.charter.com) |
| 08:47.11 | *** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr) |
| 08:52.03 | donatas | jkroon: yes |
| 08:52.42 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:53.42 | donatas | what is wrong here ? http://pastebin.com/m6fdd115e |
| 08:53.46 | donatas | i have a loop |
| 08:54.16 | jkroon | so unloop it :p |
| 08:54.49 | jkroon | for one, you have two t extensions in the same context. |
| 08:54.51 | donatas | how ? :) |
| 08:55.09 | donatas | sorry the last should be "i" |
| 08:56.16 | jkroon | also just check but it looks like you're jumping to 1@ivr:1 which as far as I can tell don't exist. |
| 08:56.40 | jkroon | log in on your CLI, type "set core verbose 4" then get into your IVR and see what it does. |
| 08:56.44 | donatas | wait, i will put all extension.conf |
| 08:57.40 | jkroon | i'm not going to read all of it. |
| 08:59.34 | donatas | hmz it timeouts.. |
| 09:00.02 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 09:00.25 | donatas | looks like http://pastebin.com/m7777d2b8 |
| 09:02.06 | jkroon | because it jumps to 1@ivr:1 which doesn't exist. |
| 09:02.32 | jkroon | open your CLI and LOOK AT WHAT ASTERISK OUTPUTS |
| 09:03.43 | donatas | where do you see 1@ivr:1 |
| 09:05.04 | *** join/#asterisk Da-Geek (n=Da-Geek@62.189.17.99) |
| 09:06.17 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 09:08.04 | jkroon | sorry, my error, you're jumping back onto itself, so after you press three it will keep saying three, three, three, three ... indefinitely. |
| 09:09.46 | donatas | can i fix it ? |
| 09:11.14 | jkroon | yes, Goto(111,1) |
| 09:11.22 | jkroon | that'll go back to the IVR, or you can Hangup() |
| 09:11.30 | jkroon | and it's not broken, it does exactly what you told it to. |
| 09:12.54 | *** join/#asterisk Jenna (n=JJ@unaffiliated/jenna) |
| 09:13.02 | donatas | don't understand why it timeouts ;\ |
| 09:13.30 | Jenna | hiya all. |
| 09:14.05 | Jenna | Is asterisk binaries at "http://www.asterisk.org/downloads/yum" production ready ? |
| 09:17.46 | Tim_Toady | jenna they should be |
| 09:18.07 | Jenna | Tim_Toady, yeah they should be. but are they ? |
| 09:18.25 | Tim_Toady | they are used in asterisk now |
| 09:18.42 | Tim_Toady | and i use them too in a couple of boxes |
| 09:18.58 | Tim_Toady | never had any major prob |
| 09:20.15 | jkroon | would a client kill a call if the SIP sequence misses a SIP/2.0 100 Trying packet? |
| 09:20.34 | jkroon | might the call continue and then 30 seconds into the call actually coming UP get dropped? |
| 09:20.41 | jkroon | http://pastebin.co.za/76928 |
| 09:21.00 | jkroon | that call got dropped, the only thing that seems to be different is that there was never a SIP/2.0 100 packet? |
| 09:21.07 | *** join/#asterisk aidinb (n=Aidin@24-176-216-154.dhcp.lnbh.ca.charter.com) |
| 09:23.28 | *** join/#asterisk ayrjola (n=ayrjola@gw.ccxtech.fi) |
| 09:25.54 | *** join/#asterisk fiddur (n=fiddur@192.121.104.121) |
| 09:26.12 | Jenna | Tim_Toady, okay |
| 09:26.45 | Jenna | Tim_Toady, btw notice this asterisknow besides other packages asterisk16, 14 etc.. what is this ? |
| 09:27.03 | Tim_Toady | asterisk 1.4.x and asterisk 1.6.x |
| 09:27.26 | Jenna | Tim_Toady, I know about asteriskNOW livecd. but what is this ? |
| 09:27.46 | ChannelZ | it's linux + asterisk + freepbx (GUI) |
| 09:28.38 | Tim_Toady | ah sorry i didnt got the question, asterisknow is a centos based distro that comes with freeppbx preinstalled |
| 09:29.18 | Jenna | IC. so if I yum install asteriskNOW . I should have everything up & running in quicktimte ? |
| 09:30.02 | *** join/#asterisk RicWright (n=Ricwrigh@firewall.a-n-t.net) |
| 09:30.07 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 09:31.14 | Tim_Toady | its not a package, its a distro |
| 09:32.48 | Jenna | Tim_Toady, so now I have clean centos 5.X installed. doing yum install asteriskNOW will result in what ? |
| 09:32.52 | jkroon | ok, asterisk (1.6.1.12) seems to in some cases not send a SIP/2.0 100 Trying packet before sending SIP/2.0 180 Ringing, this seems to be what's causing my 30 second calls. |
| 09:34.43 | jkroon | is this something that should be fixed or is this a feature? |
| 09:34.48 | Tim_Toady | jenna im not sure, never tried it, actually i just noticed thats thers a package named astersknow |
| 09:35.28 | ChannelZ | as a package it might just be asterisk+freepbx as a unit |
| 09:35.34 | Tim_Toady | propably |
| 09:41.30 | *** join/#asterisk Dovid (n=annon@109.160.160.251) |
| 09:41.32 | Dovid | j #linux |
| 09:41.42 | ChannelZ | bzzzt try again |
| 09:43.44 | ChannelZ | How did Qwest get q.com? |
| 09:44.46 | TommyBotten | Early on I assume |
| 09:45.16 | TommyBotten | During the first years you could do pretty much anything ;) |
| 09:45.18 | ChannelZ | unless the records got fudged, it says 2008 |
| 09:45.27 | creativx | 1 letter |
| 09:45.31 | creativx | impressive |
| 09:45.43 | ChannelZ | yeh all the others are IANA reserved |
| 09:45.46 | *** join/#asterisk chendy (n=chatzill@204.152.211.137) |
| 09:45.51 | TommyBotten | Well, it Creation Date: 30-mar-1999 |
| 09:45.54 | TommyBotten | eh... says |
| 09:45.57 | ChannelZ | (well I didn't check them _all_) |
| 09:46.20 | TommyBotten | But it is probably just an update of the domain, even in 1999 |
| 09:46.28 | ChannelZ | Oh yes, looked at the lower registrar date |
| 09:46.38 | TommyBotten | http://en.wikipedia.org/wiki/Single-letter_second-level_domain |
| 09:46.42 | Dovid | u need to kiss ass to get it ;) |
| 09:47.29 | ChannelZ | apparently. Qwest are so f*ing clueless about everything else I tend to doubt they had the real presence of mind to register it way-back-when |
| 09:47.48 | TommyBotten | It was registered in 92 |
| 09:47.57 | TommyBotten | Read the article :) |
| 09:48.03 | TommyBotten | interesting stuff |
| 09:48.49 | ChannelZ | ah.. so they bought it off someone else |
| 09:50.06 | ChannelZ | they didn't exist until 96 |
| 09:50.10 | dudee | need some assistance. i have an issue. i dont think its a dial plan. both analog phones hooked up to asterisk using tdm2400p. problem is when trying to dial internal extension to internal extension, it takes 5 seconds before attempting to ring the extension. i dont have this issue for external to internal or from internal to external. only internal extension to internal extension. e.g. extension 551 calls extension 552, it takes 5 seconds before it even rings. |
| 09:50.11 | *** join/#asterisk c0rnoTa (n=c0rnoTa@188.162.29.116) |
| 09:50.31 | ChannelZ | dudee: I answered this an hour ago |
| 09:50.41 | *** part/#asterisk c0rnoTa (n=c0rnoTa@188.162.29.116) |
| 09:50.53 | ChannelZ | dudee: Or gave a plausible cause anyway, you never came back with anything more specific (like your dialplan) |
| 09:51.49 | ChannelZ | ..well more than an hour ago.. |
| 09:52.04 | dudee | ChannelZ: my apologies, i was on the phone with a client |
| 09:52.07 | dudee | just came off |
| 09:52.22 | dudee | i just scrolled up to read it |
| 09:52.49 | dudee | the problem is i have an IVR, and if i dial through the IVR, it transfer instantly |
| 09:53.07 | dudee | but for internal extensions to internal, it takes 5 seconds to ring |
| 09:53.20 | jkroon | dudee, open your CLI, set the verbosity to 4, and then dial it, see where it gets stuck. |
| 09:53.43 | ChannelZ | perhaps you have other included contexts internally that aren't visible through your IVR.. but we're not mind-readers |
| 09:53.56 | jkroon | why does sip debug get sent to verbose btw? |
| 09:55.22 | ChannelZ | dunno, but it certainly is verbose |
| 09:55.22 | *** join/#asterisk corruptor (n=corrupto@85.94.32.98) |
| 09:56.29 | dudee | it does ring, but takes a while |
| 09:56.32 | dudee | <PROTECTED> |
| 09:56.32 | dudee | <PROTECTED> |
| 09:56.32 | dudee | <PROTECTED> |
| 09:56.32 | dudee | <PROTECTED> |
| 09:56.32 | dudee | <PROTECTED> |
| 09:56.36 | dudee | thsts initial |
| 09:56.45 | ChannelZ | don't do that |
| 09:56.50 | ChannelZ | http://pastebin.ca |
| 09:56.57 | dudee | my apologies |
| 09:57.01 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 09:57.36 | ChannelZ | re: pastebin your extensions.conf |
| 09:59.58 | jkroon | dudee, also, if you send it to a file with logger.conf it gets timestamps too - which is useful seeing that your problem is a 5 second delay - and we'd like to see at which step it happens. |
| 10:00.08 | dudee | ChannelZ: i know you not going to like my next line. its messy coz of freepbx and i know you going to ask me to go there but no one is currently helping there |
| 10:00.27 | jkroon | dudee, and just mentioning freepbx here gets you sent to #freepbx :) |
| 10:01.04 | Dovid | lol |
| 10:01.49 | dudee | lol |
| 10:01.55 | dudee | see thats what i mean |
| 10:01.59 | Dovid | i know some1 that was trying to hire and "asterisk expert". a guy showed up. he said show me how you install asterisk, the guy installed trixbox. he said thats not asterisk the guy coming down said trixbox == Asterisk. how sad |
| 10:02.08 | ChannelZ | Is it really 5 seconds or is that just a guess? |
| 10:02.24 | jkroon | sounds like a timeout of lookup or something. |
| 10:02.31 | dudee | around 5-6 seonds |
| 10:02.37 | dudee | *seconds |
| 10:02.41 | Dovid | that or DNS issues ;) |
| 10:02.51 | Dovid | see how long it taked to do a ping to say google.com |
| 10:02.55 | Dovid | takes* |
| 10:03.03 | Dovid | how long for the DNS to come back |
| 10:03.15 | dudee | Dovid: thanks, but i dont think so |
| 10:03.16 | ChannelZ | well yeah it could be any number of things. |
| 10:03.32 | Dovid | dudee: don't hurt to tru |
| 10:03.34 | ChannelZ | A match-as-you-go problem is usually like 3 seconds I think |
| 10:03.35 | Dovid | try* |
| 10:03.52 | dudee | i get where this is going |
| 10:03.55 | dudee | pastebin |
| 10:04.41 | dudee | will you guys be around, i just have to attend to one interviewee , should be 20 mins |
| 10:04.52 | ChannelZ | is going to bed |
| 10:04.59 | dudee | thanks |
| 10:05.06 | jkroon | yea, we had a client complaining about a 60 second delay (ok, a client of mine's client ... just to be clear), so when we got to the box it was so badly set up dns was sent to non-existing servers, shees, it was bad. |
| 10:05.08 | ChannelZ | good luck |
| 10:05.45 | jkroon | took us two days to sort out all the crap. |
| 10:06.04 | Dovid | lol |
| 10:06.17 | dudee | jkroon: no sip involved for now, i dont see dns being involved |
| 10:06.26 | Dovid | doesent matt4er |
| 10:06.38 | Dovid | dns still creates issues |
| 10:06.41 | jkroon | half of it was elastix itself, half of it was bad config settings. |
| 10:06.43 | Dovid | even if u arnet using sip |
| 10:06.52 | Dovid | i have a customer that has elastix |
| 10:06.57 | Dovid | spent time debuging last night |
| 10:06.58 | jkroon | and yea, dovid is right, any lookup of any sorts can require dns indirectly. |
| 10:07.03 | dudee | Non-authoritative answer: |
| 10:07.03 | dudee | www.google.com.sg canonical name = www.google.com. |
| 10:07.03 | dudee | www.google.com canonical name = www.l.google.com. |
| 10:07.03 | dudee | Name: www.l.google.com |
| 10:07.03 | dudee | Address: 216.239.61.104 |
| 10:07.15 | Dovid | how long does it take to resolve |
| 10:07.16 | dudee | it resolves instantly |
| 10:07.18 | jkroon | tcpdump is also a useful tool. |
| 10:07.33 | Dovid | then it's not dns |
| 10:07.43 | jkroon | tcpdump -i eth0 in one term, ast cli in another, then let rip. |
| 10:07.44 | Dovid | look at the logs |
| 10:07.49 | jkroon | see what traffic goes when and where. |
| 10:07.56 | jkroon | logs are always a good starting point. |
| 10:08.04 | *** join/#asterisk mrchrisadams (n=chrisada@87-194-125-43.bethere.co.uk) |
| 10:08.05 | ChannelZ | Do you use realtime config? MySQL logging? we could sit here all night listing 150 things it COULD be |
| 10:08.15 | dudee | i have a feeling it's timeout issue. as if it's waiting |
| 10:08.15 | jkroon | Dovid, can still be, if the NS servers for the destination is non-responsive ... |
| 10:08.24 | jkroon | TIMEOUT OF WHAT?!? |
| 10:08.32 | dudee | thats the question |
| 10:08.40 | dudee | it's internal extension to internal extension |
| 10:08.50 | jkroon | NO. that's the symptom. |
| 10:08.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 10:09.04 | ChannelZ | turns on the echo chamber |
| 10:09.11 | ChannelZ | pastebin your extensions.conf |
| 10:09.21 | jkroon | the dialplan is doing something which requires some lookup, and that LOOKUP is timing out. |
| 10:09.28 | jkroon | ChannelZ, no point, it's freepbx :p |
| 10:09.41 | dudee | jkroon: thanks for that |
| 10:10.15 | ChannelZ | dudee: what happens if you configure a softphone, in the same context as your 'internal' phones? |
| 10:11.55 | mrchrisadams | hi guys, is the AsteriskGUI project essentially freepbx now? |
| 10:12.02 | dudee | soft phone instant |
| 10:12.20 | ChannelZ | ok so as I said it's more than likely a match-as-you-go problem.. |
| 10:12.32 | tzafrir_laptop | mrchrisadams, no. AsteriskNOW user FreePBX |
| 10:12.40 | mrchrisadams | I'm new to Asterisk as of two days ago, and I'm struggling to make sense of the history, and where switchvox, freepbx and the like fit in together |
| 10:12.46 | tzafrir_laptop | asterisk-gui is a different (and smaller) beast |
| 10:12.51 | ChannelZ | there is another extension that COULD match the one you are dialing, if you were to dial more digits |
| 10:12.56 | dudee | ChannelZ: so how do i diagnose it to rectify or better ir? |
| 10:13.25 | dudee | i guess maybe the outbound trunks |
| 10:13.30 | dudee | that might be a good idea |
| 10:13.34 | ChannelZ | you have to look at your dialplan/extensions and figure that out yourself since we're not going to get to see it apparently |
| 10:13.40 | dudee | i use NXXXXXXX |
| 10:13.47 | dudee | and the internal is 5XX |
| 10:13.53 | ChannelZ | ok well there you go |
| 10:14.05 | dudee | thanks |
| 10:14.06 | dudee | let me try |
| 10:14.09 | mrchrisadams | tzafrir_laptop: my understanding (so far at least) is that freepbx is the largest free GUI system around for Asterisk, but switchvox is the commercial option digium are spending more time focussing on developing, right? |
| 10:14.18 | ChannelZ | 5XX is inclusive of NXXXXXXX |
| 10:14.32 | tzafrir_laptop | mrchrisadams, I can't comment for Digium |
| 10:14.38 | ChannelZ | so you type 512 and * is waiting to see if you might dial 512837483 |
| 10:14.40 | ChannelZ | or whatever |
| 10:14.51 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 10:15.01 | tzafrir_laptop | I can't really work with software that is not free |
| 10:16.28 | mrchrisadams | tzafrir_laptop: do you use AsteriskNOW? I'm looking for the simplest system possible to start working with, to get an understanding of it, and AsteriskNow looks like it, but I might be missing something obvious |
| 10:16.43 | dudee | ChannelZ: great thanks |
| 10:16.47 | dudee | you are great |
| 10:16.55 | dudee | now its 2 seconds |
| 10:16.58 | dudee | i can live with that |
| 10:17.01 | dudee | caller id i assume |
| 10:17.03 | dudee | but alright |
| 10:25.28 | jkroon | ok, add a Progress() call into the dialplan before Dial(SIP/phone) and all is working happy now. Same on outbound calls before Dial(...) and everything is AOK it seems. |
| 10:26.11 | jkroon | shouldn't calling Dial() force a SIP/2.0 100 Trying packet to be sent if it hasn't already? or what exactly is the meaning of all of those SIP packets? |
| 10:27.09 | angryuser_ | jkroon, that sip packets letting you to call |
| 10:27.30 | jkroon | yea thanks angryuser_ ... that's useful. |
| 10:27.59 | jkroon | i'm trying to understand what exactly 100 Trying and 186 Progress actually means. |
| 10:28.18 | jkroon | if i were a phone and i were to receive those - how should I react? what is the peer trying to tell me? |
| 10:28.23 | angryuser_ | jkroon, well, you waited for a complete course about sip protocol here ? Google "Sip protocol" |
| 10:29.05 | angryuser_ | 100 > trying, Trying to reach the destination but not sure > 186 In progress of reaching destinations, pretty self explanatory |
| 10:31.18 | jkroon | so a typical headset would typically ignore those packets? |
| 10:32.16 | jkroon | nm. i'll go read a book. |
| 10:32.20 | *** part/#asterisk jkroon (n=jkroon@dsl-244-25-144.telkomadsl.co.za) |
| 10:33.38 | ChannelZ | sounds prudent |
| 10:34.44 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 10:41.04 | *** join/#asterisk Heretic (n=fallen@dsl-246-89-133.telkomadsl.co.za) |
| 10:48.06 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
| 10:56.56 | *** join/#asterisk jtodd (i=gm6ljr9o@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 10:56.56 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 11:02.05 | *** join/#asterisk waa (n=waa@balrog.credipar.com.br) |
| 11:27.06 | *** join/#asterisk cuco (n=Diego@local.xorcom.com) |
| 11:30.27 | Jenna | is there a special way to include a file in chan_dahdi.conf . Im getting this error. http://pastebin.ca/1750845 |
| 11:31.14 | Jenna | this is the line that I added on the top of chan_dahdi.conf #include "/etc/asterisk/dahdi-channels.conf" |
| 11:39.55 | Jenna | anyone around ? |
| 11:40.49 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 11:42.12 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 11:43.18 | Dovid | nope |
| 11:44.25 | Dovid | jenna: I believe it should be #include /etc/asterisk/dahdi-channels.conf |
| 11:44.32 | Dovid | without the quotes |
| 11:44.37 | *** join/#asterisk jbw (n=jbw@dsl-105-162.cust.imagine.ie) |
| 11:44.53 | Jenna | hmm. yup but it still coughs up that error |
| 11:45.04 | Jenna | http://pastebin.ca/1750845 |
| 11:45.37 | Dovid | i cant seem to get to pastebin.ca |
| 11:45.59 | Dovid | can u do pastebin.com ? |
| 11:46.13 | Jenna | [Jan 10 08:44:50] WARNING[3538]: config.c:1089 process_text_line: parse error: No category context for line 12 of /etc/asterisk/dahdi-channels.conf |
| 11:46.18 | Jenna | sure wait |
| 11:46.25 | TommyBotten | Jenna: You have a syntax error |
| 11:46.31 | Dovid | duh ! |
| 11:46.47 | Dovid | the issue could be in /etc/asterisk/dahdi-channels.conf |
| 11:46.51 | Dovid | can you post that as well ? |
| 11:46.52 | TommyBotten | Jenna: Could you please paste your config file aswell? |
| 11:47.04 | Jenna | that was generated by dahdi_genconf |
| 11:47.12 | Jenna | sure wait |
| 11:47.24 | Dovid | and post on pastebin.com |
| 11:47.43 | *** join/#asterisk Slashman (n=Slash@ariane.fimasys.com) |
| 11:47.45 | Dovid | for some reason my ISP is having issues to pastebin.ca and i am too lazy to set up a tunnel to one of my other servers |
| 11:48.21 | Jenna | http://pastebin.com/m1636e511 |
| 11:48.47 | Dovid | s[Jan 10 08:44:50] ERROR[3538]: config.c:1081 process_text_line: The file '/etc/asterisk/dahdi-channels.conf' was listed as a #include but it does not exist. |
| 11:48.59 | Dovid | what happens when you do: cat /etc/asterisk/dahdi-channels.conf |
| 11:50.21 | Jenna | Dovid, :) trust me its there. with owner ship changed to asterisk:asterisk as well |
| 11:51.05 | Dovid | i know it is. but for some reason asterisk does not see that. y do you do an include and not put it directly in to chan_dahdi.conf ? |
| 11:51.13 | Jenna | does the sequence of #include /etc/asterisk/dahdi-channels.conf in the file chan_dahdi.conf matter ? i.e. top of the file or end of the file |
| 11:51.26 | Dovid | it should not matter where it is |
| 11:51.40 | TommyBotten | It should not, but in most cases it is at the bottom |
| 11:51.45 | Dovid | the error asterisk is throwing is that it does not see the file at all. could be a permisuins issue |
| 11:51.55 | Dovid | try chmod 644 dahdi-channels.conf |
| 11:53.13 | Jenna | okay |
| 11:54.25 | *** join/#asterisk ruyo (n=psantos@195.23.253.223) |
| 11:55.10 | Jenna | nope no luck |
| 11:56.27 | TommyBotten | strace it so you can see each file asterisk tries to access |
| 11:56.32 | TommyBotten | That is, if you are insane |
| 11:56.52 | Jenna | guys putting that file in the end did the trick |
| 11:57.01 | Jenna | but with one warning WARNING[3891]: chan_dahdi.c:4676 handle_alarms: Detected alarm on channel 2: Red Alarm |
| 11:57.06 | TommyBotten | Ah |
| 11:57.14 | TommyBotten | Now you can communicate with the card at least |
| 11:57.40 | Jenna | yup dahdi show channels work now |
| 11:58.08 | Jenna | thanx. but it was a weird errr thing. |
| 12:07.23 | Dovid | did it work after the chmod ? |
| 12:13.14 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 12:15.11 | *** join/#asterisk ChrisWi (n=admin@mx2.wwserver.net) |
| 12:15.35 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 12:16.01 | shamelessn00b | hi guys, I need a little help on perl fastagi server configuration and scripting anyonearound?? |
| 12:16.50 | *** join/#asterisk Akiraa (n=Akira@92.81.205.184) |
| 12:18.01 | mrchrisadams | hi guys - anybody here know if there's a copy of the stencils network diagrams used in the oreilly asterisk book around at all? |
| 12:23.25 | Dovid | i don't know perl :( |
| 12:23.29 | Dovid | whats the exact issue |
| 12:23.48 | shamelessn00b | I installed the perl FastAGI server module |
| 12:24.07 | shamelessn00b | I want to know how do I make a connection to that server from my asterisk box |
| 12:24.32 | shamelessn00b | there must be some deamon I should start on the machine I am running the FastAGIserver on |
| 12:25.19 | Dovid | you want to know how to start it or how to connect from asterisk ? |
| 12:26.29 | shamelessn00b | both |
| 12:26.51 | shamelessn00b | http://search.cpan.org/~jaywhy/Asterisk-FastAGI-0.02/lib/Asterisk/FastAGI.pm |
| 12:26.57 | shamelessn00b | im on this page at the moment |
| 12:27.12 | shamelessn00b | and Im not quite getting it |
| 12:36.14 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:36.14 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 12:42.15 | shamelessn00b | anyone using FastAGI |
| 12:42.19 | shamelessn00b | perl |
| 12:42.22 | shamelessn00b | ? |
| 12:42.37 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 12:42.55 | leifmadsen | shamelessn00b: many people are -- you may wish to ask a more specific question |
| 12:44.16 | shamelessn00b | Hi leifmadsen , I'd like to know how exactly do I configure a FastAGI server and start sending AGIs to it to execute |
| 12:44.19 | shamelessn00b | from asterisk |
| 12:44.50 | shamelessn00b | I've Configured Asterisk-FastAGI-0.02 package on one of my machines |
| 12:45.17 | *** join/#asterisk jkroon (n=jkroon@dsl-244-25-144.telkomadsl.co.za) |
| 12:45.24 | shamelessn00b | http://search.cpan.org/~jaywhy/Asterisk-FastAGI-0.02/lib/Asterisk/FastAGI.pm I'm following this page |
| 12:45.57 | shamelessn00b | I've never really used AGIs before so I don't know what to do |
| 12:51.04 | shamelessn00b | leifmadsen: there?? |
| 12:51.12 | leifmadsen | shamelessn00b: no, I'm currently working |
| 12:51.15 | leifmadsen | and I don't use fastAGI |
| 12:51.30 | shamelessn00b | ok |
| 12:54.41 | corruptor | heh it seems that 1.6.0 svn is broken. seg faults on reload |
| 12:58.26 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 12:59.06 | *** join/#asterisk freckle (n=jon@195.74.96.122) |
| 12:59.49 | *** join/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 13:06.10 | Delido1983 | Hello, i have an problem with an group Pickup. I don´t use the pickup feature in the features.conf i want to write an extention for this. Example: exten => *8/440,1,PickUp(414&449&453&455&467&471&498&499) <- We have pri and sip phones. By Sip-to-sip calls the Pickup extension is okay but if an dahdi call come in to a sip phone the log show by taking a pickup: NOTICE[21516]: app_directed_pickup.c:159 pickup_exec: No target channel found for. |
| 13:07.01 | ManxPower-work | Delido1983: Try SIP/414&SIP/449, etc |
| 13:07.53 | Delido1983 | Executing [*8@DLPN_IT:1] Pickup("SIP/485-00002c93", "SIP/486&SIP/482") in new stack [Jan 14 14:07:36] NOTICE[21661]: app_directed_pickup.c:159 pickup_exec: No target channel found for SIP/486. [Jan 14 14:07:36] NOTICE[21661]: app_directed_pickup.c:159 pickup_exec: No target channel found for SIP/482. |
| 13:07.57 | ManxPower-work | Delido1983: Remember Pickup picks up channels, not extensions |
| 13:09.55 | Delido1983 | but how can i realize that? |
| 13:10.35 | Delido1983 | i dont want use the feature and not the puckupgroup, callgroup entry in the users.conf |
| 13:10.48 | Delido1983 | pickupgroup |
| 13:11.24 | Delido1983 | because our pickup rule are complex |
| 13:11.50 | ManxPower-work | Delido1983: I can't help you with users.conf. Looks like Pickup does pick up extensions, not channels. (at least in 1.6.x). Are all these extensions 414&449&453&455&467&471&498&499 all in the same extensions.conf context? |
| 13:13.06 | ManxPower-work | i.e. are those extensions defined in the [DLPN_IT] in extensions.conf? |
| 13:13.33 | Delido1983 | yes all Phones are in the extension callingrule_mitarbeiter |
| 13:13.50 | Delido1983 | an incoming call are in the context DID_span_1 |
| 13:14.01 | ManxPower-work | well if those extensions are in [callingrule_mitarbeiter] then trying to pick them up in a different context won't work. |
| 13:14.05 | *** join/#asterisk neurosys (n=neurosys@173-9-159-182-miami.txt.hfc.comcastbusiness.net) |
| 13:14.43 | Delido1983 | if an call come in this show core show channels: tk*CLI> core show channels Channel Location State Application(Data) SIP/414-00002c79 (None) Up AppDial((Outgoing Line)) DAHDI/31-1 s@macro-eingehend:6 Up Dial(SIP/414|85|tT) |
| 13:15.13 | ManxPower-work | Delido1983: do you understand how contexts work in Asterisk? |
| 13:15.49 | Delido1983 | i think yes.. but im confused :D |
| 13:15.55 | ManxPower-work | yes, you are confused. |
| 13:16.24 | ManxPower-work | Pickup is not finding the extension you specified in the current context. |
| 13:20.23 | Delido1983 | mhm okay thanks |
| 13:21.36 | *** part/#asterisk fiddur (n=fiddur@192.121.104.121) |
| 13:24.36 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 13:27.31 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 13:28.10 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:29.15 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:29.45 | jaytee | morning [TK]D-Fender |
| 13:29.51 | [TK]D-Fender | jaymornin' |
| 13:30.14 | *** join/#asterisk e4 (n=e4@rrcs-76-79-59-194.west.biz.rr.com) |
| 13:30.35 | jaytee | we're having a heat wave! it's expected to get up to at least 40F here today |
| 13:33.55 | *** part/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 13:35.52 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 13:38.23 | ManxPower-work | jaytee: Same here. We had snow on the ground for 5 days. That's not supposed to happen in Huntsville, AL |
| 13:38.35 | jblack | jaytee: lucky bastard. 34 predicted here. |
| 13:39.15 | *** join/#asterisk uluatu (n=uluatu@200.146.5.244.dynamic.dialup.gvt.net.br) |
| 13:40.29 | *** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com) |
| 13:40.56 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:40.56 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:41.40 | fors1 | anyone experiencing segfaults in chan_iax2.so when rtchachefriends=yes? Asterisk 1.6.2.0 |
| 13:41.53 | fors1 | anyone else that is |
| 13:44.33 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 13:44.48 | *** join/#asterisk ickmund (n=magnus@ada-bcn-fw01.adamoeurope.com) |
| 13:45.00 | shamelessn00b | can I use my existing AGIs and use them with FastAGI (Perl) |
| 13:45.02 | shamelessn00b | ?? |
| 13:45.32 | ManxPower-work | shamelessn00b: That seems like it's pretty easy to look up. |
| 13:46.08 | shamelessn00b | ManxPower-work: problem is, I've never used AGIs |
| 13:46.10 | shamelessn00b | so I dont know a thing about them |
| 13:46.18 | ManxPower-work | shamelessn00b: then don't start off with FastAGI |
| 13:46.54 | shamelessn00b | I've joined this company |
| 13:47.01 | shamelessn00b | and they are running many AGIs already |
| 13:47.15 | shamelessn00b | they want to optimize server performance by using FastAGI |
| 13:47.31 | ManxPower-work | shamelessn00b: then you must modify the existing AGIs |
| 13:47.54 | [TK]D-Fender | shamelessn00b: BOOK <_ |
| 13:47.59 | [TK]D-Fender | ~book |
| 13:48.00 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 13:48.04 | shamelessn00b | that was what I was asking, that would I have been able to use existing AGIs without modification |
| 13:48.15 | ManxPower-work | [TK]D-Fender: odd, how we keep telling him to look at the book. It's almost like he didn't read it or something. |
| 13:48.17 | [TK]D-Fender | shamelessn00b: And the interface is different |
| 13:48.27 | ManxPower-work | shamelessn00b: it took me 3.5 seconds to find that information on Google. |
| 13:48.47 | shamelessn00b | ManxPower-work: you knew what to look for |
| 13:49.03 | shamelessn00b | I usually dont ask questions if I have some idea what to look for |
| 13:49.09 | ManxPower-work | shamelessn00b: I searched for "agi fastagi". |
| 13:49.51 | ManxPower-work | shamelessn00b: there are AGI and FastAGI examples in the Asterisk source tarball |
| 13:49.54 | shamelessn00b | I've already visited the first three links |
| 13:49.59 | shamelessn00b | of that result |
| 13:50.14 | *** join/#asterisk dennis00 (i=dennis00@unaffiliated/dennis00) |
| 13:50.20 | jkroon | what's the policy on making changes to accomodate specific software? CounterPath's soft clients drops calls 30 seconds after being answered if they don't receive both a 100 Trying AND a 180 Ringing before the call is answered, and this doesn't always happen. |
| 13:50.25 | shamelessn00b | and 6th one as well |
| 13:50.32 | dennis00 | Hi, my phone calls stop when I press #? Why is that? |
| 13:51.12 | [TK]D-Fender | dennis00: Sorry, could you be a little more vague please? |
| 13:51.14 | dennis00 | Unfortunately googling asterisk and number sign did not help much. |
| 13:51.35 | ManxPower-work | dennis00: I suspect you are using an Asterisk GUI. |
| 13:51.37 | jkroon | I've got SIP traces for both cases. I have emailed them regarding it but am not expecting a positive response. |
| 13:51.39 | dennis00 | [TK]D-Fender: Definitely. When I call, I come in a stupid menu, which tells me to put in my bank number. |
| 13:52.00 | dennis00 | [TK]D-Fender: After that, to confirm, I must press #, but then some voice comes up, asking me if I want to make a new call. |
| 13:52.20 | [TK]D-Fender | dennis00: you have enabled DTMF transfers <- |
| 13:52.21 | ManxPower-work | jkroon: I doubt they will include code to fix problems with a specific application. In your dialplan do an Answer before doing anything else. This is normally a bad thing, but if you need it to work around crappy softphones then you need it. |
| 13:52.24 | dennis00 | ManxPower-work: No, I decided not to use a GUI. FreePBX was a disapointment. |
| 13:52.43 | ManxPower-work | dennis00: then remove the t/T/w/W/k/K options from your Dial line. |
| 13:52.44 | dennis00 | [TK]D-Fender: Well, the numbers did work, I had to press, 1, 2, then 324823#, suppose that was my bank nr. |
| 13:53.04 | dennis00 | Let me see |
| 13:54.11 | dennis00 | I have not changed either of those settings. Should I do both? |
| 13:54.27 | ManxPower-work | dennis00: "changed"? |
| 13:54.55 | dennis00 | Yes, I have not set those settings. |
| 13:55.02 | ManxPower-work | dennis00: paste the output of a failed call to pastebin.ca |
| 13:55.04 | dennis00 | But I think DTMF already works, doesn't it? It works for numbers... |
| 13:55.18 | ManxPower-work | dennis00: yes. you have endabled DTMF transfers to be initiated by a # |
| 13:55.38 | dennis00 | I will paste in pastebin.ca. |
| 13:56.36 | dennis00 | Please enter the telephone number without zero zero, that's what it says. |
| 13:56.41 | dennis00 | Let me change the debug settoins, 1 sec. |
| 13:56.50 | ManxPower-work | dennis00: Asterisk does not talk unless you tell it to. |
| 13:57.23 | ickmund | ManxPower-work, I wish people were the same... *sigh* |
| 13:57.27 | ManxPower-work | I suspect you have GUIfied files laying around everywhere, but I'll know when you pastebin. |
| 13:58.48 | dennis00 | I don't use a GUI. |
| 13:58.56 | ManxPower-work | dennis00: less talk, more pastebin |
| 13:59.00 | dennis00 | But I cannot find how I can increase verbosity level. |
| 13:59.15 | ManxPower-work | dennis00: asterisk -rvvv is all the verbosity I should need. |
| 13:59.35 | ManxPower-work | this isn't rocket science. I think I know exactly what your problem is, but I need that pastebin. |
| 14:00.02 | ManxPower-work | I don't need debug, I don't need verbosity set to a million. |
| 14:00.34 | dennis00 | no extra details. |
| 14:00.41 | dennis00 | In asterisk -rvvv. |
| 14:00.49 | dennis00 | when pressing codes. |
| 14:01.03 | ManxPower-work | Does it show a line with the word Dial in the console? |
| 14:01.15 | dennis00 | Yes |
| 14:01.17 | ManxPower-work | dennis00: you do not know enough to know what I'm looking for. |
| 14:01.32 | dennis00 | But while connected it does not give any additional lines. |
| 14:01.41 | dennis00 | e.g. when pressing keys for a menu. |
| 14:01.45 | ManxPower-work | dennis00: IT'S NOT SUPPOSED TO |
| 14:01.58 | ManxPower-work | now either provide the pastebin or someone else can help you. |
| 14:02.16 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 14:02.27 | dennis00 | http://pastebin.ca/1750969 I hope this is enough. |
| 14:03.15 | ManxPower-work | dennis00: Who is your carrier? |
| 14:03.24 | dennis00 | ManxPower-work: Budgetphone. |
| 14:03.34 | dennis00 | A Dutch provider. |
| 14:03.43 | ManxPower-work | dennis00: it appears they are the ones capturing the # |
| 14:03.57 | dennis00 | Hmm,, let me give them another call. |
| 14:04.16 | ManxPower-work | dennis00: just to be safe, rename /etc/asterisk/features*.conf to something else, restart (not reload) asterisk and try it again |
| 14:04.48 | *** join/#asterisk etfonhomey (n=etfonhom@ip-64-32-192-35.iad.megapath.net) |
| 14:04.56 | dennis00 | ok, calling. |
| 14:05.27 | ManxPower-work | dennis00: that pastebin confirms what you have told us. You'd be surprised at how often someone tells us something that does not match the CLI output they provide. |
| 14:05.51 | dennis00 | I see, well, I appreciate your help. |
| 14:05.59 | dennis00 | It says 'please enter the telephone number' again after the #. |
| 14:06.42 | ManxPower-work | dennis00: *nod* without a /etc/asterisk/features*.conf Asterisk would never ever play that message when you press # |
| 14:06.52 | dennis00 | ohhhh |
| 14:06.53 | ManxPower-work | So the issue must be somewhere else (I suspect the probider) |
| 14:07.07 | dennis00 | I renamed it to features.conf_, that's ok right? |
| 14:07.16 | ManxPower-work | dennis00: should be ok |
| 14:07.36 | dennis00 | thank you. |
| 14:07.49 | [TK]D-Fender | dennis00: where do Is ee * saying what yous ay it does? |
| 14:08.08 | [TK]D-Fender | dennis00: where do I see * saying what you say it does? |
| 14:08.32 | dennis00 | *? in features*.conf? it means it loads all files starting with features and ending with .conf. |
| 14:08.39 | dennis00 | I might not understand what you're saying. |
| 14:09.06 | ManxPower-work | dennis00: Lazy people use "*" to mean "Asterisk" |
| 14:09.17 | ManxPower-work | I never do that as it can lead to confusion |
| 14:09.32 | dennis00 | Oh, it led to confusion. |
| 14:09.57 | dennis00 | [TK]D-Fender: You mean in the pastebin? The pastebin gives no info, as it should be according to ManxPower-work. |
| 14:10.15 | ManxPower-work | dennis00: it gave me lots of information. |
| 14:10.42 | dennis00 | I have tried voipbuster, it does accept the #. |
| 14:10.48 | dennis00 | via my own Asterisk server. |
| 14:11.08 | [TK]D-Fender | dennis00: * is not playing that transfer... your PROVIDER is |
| 14:11.09 | ManxPower-work | dennis00: so it works with voipbuster and not with budgetphone? |
| 14:11.20 | dennis00 | ManxPower-work: correct. |
| 14:11.21 | [TK]D-Fender | dennis00: THEY are using Asterisk, and SCREWED UP |
| 14:11.26 | ManxPower-work | dennis00: then it is your provider. |
| 14:11.45 | dennis00 | Well, let's hear what they've got to say. |
| 14:12.07 | dennis00 | It seems they make their money by long phone queues ^^ |
| 14:12.10 | dennis00 | Although I am not sure it costs money, as they use VOIP too. |
| 14:12.30 | [TK]D-Fender | .... |
| 14:13.17 | dennis00 | I am on the line. |
| 14:13.23 | dennis00 | That reminds me of a good song. |
| 14:13.41 | dennis00 | on the line might be Dutch though and not correct English. |
| 14:13.49 | Katty | wibbles |
| 14:14.48 | [TK]D-Fender | dennis00: has many meanings in english |
| 14:15.08 | dennis00 | [TK]D-Fender: a drugs-related meaning too? |
| 14:15.18 | [TK]D-Fender | dennis00: Not that I know of |
| 14:15.25 | dennis00 | [TK]D-Fender: I see. |
| 14:15.27 | dennis00 | Well, they changed it |
| 14:15.41 | ManxPower-work | I love the Nederlands |
| 14:15.49 | *** join/#asterisk jamesh1 (n=jhenders@xob.neospire.net) |
| 14:16.15 | Katty | wobbles |
| 14:16.31 | dennis00 | it's ok XD |
| 14:17.50 | [TK]D-Fender | Katty may wobble......BUT SHE WON'T FALL DOWN! |
| 14:18.59 | dennis00 | a female in this channel, we should cherish that. |
| 14:19.35 | TommyBotten | Takker |
| 14:19.45 | TommyBotten | Eh... wrong window :> |
| 14:22.26 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:22.35 | jkroon | ManxPower-work, thanks for the advice, and yea, I would really prefer to not do that. In this case I need a way to FORCE a 100 Trying (currently done by running Progress()) and then I need 180 Ringing (Ringing()) and only then should it be answered. First prize is obviously getting Counterpath to fix their crap. |
| 14:24.25 | dennis00 | My VSP also tells me that my server is blocking incoming calls, but I see no such thing in asterisk. |
| 14:24.26 | [TK]D-Fender | jkroon: No first prize goes out to the person who stops using shit soft-phones :p |
| 14:24.27 | areay | asterisk crashed, refused to be killed and to get it to start i had to restart my server... which logfile should i be looking in to find the source of the problem? and how can i *actually* kill the process if it happens again without having to reboot? (i tried killall, kill -9) ...i'm using 1.6 btw |
| 14:24.40 | *** join/#asterisk Skeeter- (i=Skeeter@c216.218.2-65.clta.globetrotter.net) |
| 14:25.14 | [TK]D-Fender | areay: safe_asterisk just reloads it automatically. You need to kill this if its running FIRST |
| 14:26.36 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:26.36 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:27.13 | dennis00 | Does the newest Asterisk not know 'set verbose'? |
| 14:27.19 | areay | [TK]D-Fender, lol i was wondering what was going on. thanks for the info |
| 14:27.25 | [TK]D-Fender | dennis00: "core set verbose" |
| 14:27.37 | [TK]D-Fender | dennis00: Read the docs. Syntax for things change between versions |
| 14:27.41 | dennis00 | [TK]D-Fender: great, thank you. |
| 14:27.47 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 14:28.21 | ManxPower-work | areay: there is no such thing as "1.6" |
| 14:28.42 | areay | ManxPower-work, 1.6.1.6 sorry |
| 14:28.49 | ManxPower-work | dennis00: read all the UPGRADE*.txt files. |
| 14:29.21 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:30.05 | jkroon | [TK]D-Fender, what soft phones do you recommend then? |
| 14:30.45 | ManxPower-work | All softphones suck |
| 14:30.58 | areay | i compiled asterisk from source (it's one of the only programs i run that i manually compiled)... what's the deal with security updates/patches, and when should i think about upgrading? i notice 1.6.2 is now available... |
| 14:31.22 | areay | i just always hear people saying not to upgrade |
| 14:31.27 | ManxPower-work | areay: read: http://www.asterisk.org/asterisk-versions |
| 14:31.32 | areay | ManxPower-work, thanks |
| 14:32.02 | drmessano | The only person I ever heard advocating not upgrading is Kerry Garrison, and I wont go into that |
| 14:34.40 | dennis00 | http://pastebin.ca/1751001 Is my inbound configuration incorrect? |
| 14:35.01 | p3nguin | I don't like upgrading, either. On the other hand, if a newer version has a critical bug fix or a new feature that you NEED, then go ahead and upgrade. If you upgrade just to see a higher version number, you're a fool. |
| 14:35.07 | Katty | dennis00: i am no different than anyone else. |
| 14:35.15 | ManxPower-work | dennis00: well you have context=dennis but you showed a different context. |
| 14:35.16 | Katty | dennis00: if it helps, picture me as a 50 year old male in the basement. |
| 14:36.35 | dennis00 | Katty: you might even be. |
| 14:36.55 | p3nguin | She is. |
| 14:36.57 | *** join/#asterisk TheDavidFactor (n=chatzill@fw1.safedataisp.net) |
| 14:36.58 | p3nguin | err... |
| 14:37.10 | dennis00 | ManxPower-work: Where do I show a different context? |
| 14:37.49 | ManxPower-work | dennis00: you have context=dennis in [1001] but above that you have [inbound] and [inbound] != [dennis] |
| 14:38.22 | drmessano | HAHAH |
| 14:38.38 | drmessano | Kerry Garrison doesn't know the difference between DVI and HDMI |
| 14:38.42 | dennis00 | [dennis] includes outbound... |
| 14:38.49 | dennis00 | but not inbound. |
| 14:39.02 | [TK]D-Fender | drmessano: one clearly has 3 charaters, and the other has 4! What a twit! |
| 14:39.15 | ManxPower-work | maybe so, but it's not in your pastebin |
| 14:39.18 | dennis00 | I didn't know it until I looked it up. |
| 14:39.23 | areay | ManxPower-work, so to upgrade, can i just 'make install' right over the top of my existing installation? obviously after stopping the asterisk service |
| 14:39.24 | *** join/#asterisk SuPrSluG (n=SuPrSluG@8.224.32.34) |
| 14:39.26 | dennis00 | ManxPower-work: I forgot the include for inbound. |
| 14:39.37 | ManxPower-work | areay: yes, after you read UPGRADE*.txt |
| 14:39.50 | leifmadsen | ~asteriskversioning |
| 14:39.51 | infobot | from memory, asteriskversioning is http://www.asterisk.org/asterisk-versions |
| 14:40.08 | leifmadsen | ~asteriskversioning16 |
| 14:40.16 | leifmadsen | hmmm, I forget what I made that one :) |
| 14:40.25 | *** join/#asterisk smooth_penguin (n=smoove@59.95.32.93) |
| 14:40.26 | leifmadsen | ~asteriskversion16 |
| 14:40.28 | dennis00 | DVI is like a no-sound HDMI, that's what I understood. |
| 14:40.33 | leifmadsen | ~asteriskversioning-1.6 |
| 14:40.36 | [TK]D-Fender | ~asterisk-versioning |
| 14:40.45 | leifmadsen | no, there was one I made for 1.6 specifically |
| 14:40.46 | dennis00 | That Kerry looks like a German friend of us. |
| 14:40.51 | leifmadsen | ~asteriskversioning1.6 |
| 14:40.57 | leifmadsen | I give up :) |
| 14:41.06 | ManxPower-work | I thought DVI and HDMI meant "better picture quality than you can perceive with your crappy vision" |
| 14:41.50 | drmessano | 888Voipstore is selling a little "mini PBX" which we found out some time ago was a rebranded micro PC they're selling as a PBX.. well, he has a demo video for the box, and in the video, he shows off the HDMI port they labeled as "DVI", which I guess was done for simplicity, since they "expect:" you to use the HDMI <> DVI adapter to connect it to a monitor |
| 14:42.09 | drmessano | Rather than noting that, he shows the adapter, and calls the DVI end "HDMI" and the HDMI end "DVI". |
| 14:42.36 | ManxPower-work | drmessano: Everyone knows all the VGA ports expired years ago. |
| 14:43.12 | drmessano | and says something about having a monitor with a standard HDMI port, and if not, use the DVI connector |
| 14:43.29 | drmessano | DUR |
| 14:43.33 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 14:44.44 | p3nguin | My TV has one jack that is an HDMI jack, and it says HDMI IN, but it is the jack to be used when I connect an HDMI cable to my computer with a DVI-HDMI adapter. |
| 14:45.05 | dennis00 | http://pastebin.ca/1751009 I updated it with my full sip.conf and extensions.conf. Incoming calls do not show up in asterisk, nor do they work. |
| 14:45.34 | p3nguin | It has three HDMI jacks, but HDMI IN 2 is to be used when a DVI adapter is in place. |
| 14:47.48 | *** join/#asterisk telnettech (n=telnette@cpe-71-74-91-116.insight.res.rr.com) |
| 14:47.52 | *** join/#asterisk girlny (n=girlny@CPE00195b4be142-CM001a668ec076.cpe.net.cable.rogers.com) |
| 14:47.58 | [TK]D-Fender | dennis00: Incoming from what? |
| 14:47.59 | p3nguin | dennis00: You don't need ulaw codec? I thought everyone used ulaw. |
| 14:48.25 | [TK]D-Fender | p3nguin: Get out of North America |
| 14:48.44 | girlny | I NEED HELP TO SET UP CDR PLEASE HELP |
| 14:48.45 | [TK]D-Fender | p3nguin: an entire planet awaits you |
| 14:48.48 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.38.141) |
| 14:48.52 | shamelessn00b | lol |
| 14:49.09 | drmessano | p3nguin: We're not talking about TVs though |
| 14:49.33 | [TK]D-Fender | dennis00: Does your * server itself have a public IP? |
| 14:50.05 | dennis00 | [TK]D-Fender: Yes it does. |
| 14:50.16 | dennis00 | [TK]D-Fender: It starts with 93. |
| 14:50.20 | ickmund | dennis00, Have you verified that the calls are actually forwarded to you? Try an ngrep on the server to see if you have any incoming packages |
| 14:50.21 | [TK]D-Fender | dennis00: then enable SIP DEBUG. basic CLI verbose = worthless |
| 14:50.23 | girlny | can someone help me to set up cdr to work in my server i been tring for 3 days and nothing |
| 14:50.43 | dennis00 | p3nguin: I allow ulaw? |
| 14:50.51 | p3nguin | dennis00: You do not. |
| 14:50.58 | dennis00 | ickmund: Yes, I called my VOIP provider. |
| 14:51.02 | [TK]D-Fender | dennis00: IGNORE. Now go enable SIPD EBUG |
| 14:52.07 | girlny | can someone help me to set up cdr to work in my server i been tring for 3 days and nothing |
| 14:52.31 | dennis00 | ok, redialing. |
| 14:52.46 | *** join/#asterisk muiro (n=muiro@unaffiliated/muiro) |
| 14:52.57 | dennis00 | well, that's a lot of lines. |
| 14:53.07 | ManxPower-work | girlny: edit /etc/asterisk/cdr.conf and make whatever changes you need. |
| 14:53.09 | tzafrir_laptop | infobot, tell girlny about ask |
| 14:53.50 | muiro | if I'm going to use realtime for say, queues, can I look in to the sample queues.conf in order to get the field names I will need? |
| 14:54.22 | dennis00 | http://pastebin.ca/1751021 |
| 14:55.11 | *** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net) |
| 14:55.21 | dennis00 | This is from VOIPBuster to BudgetPhone (010 number, my VOIP Provider) |
| 14:55.28 | girlny | i dont have /etc/asterisk/cdr.conf i have cdr_mysql.conf |
| 14:56.13 | ManxPower-work | girlny: If you want regular CDRs, those are configured in /etc/asterisk/cdr.conf. If you are trying to use MySQL for CDRs, well you didn't tell us that in the first place or I never would have responded to you. |
| 14:56.24 | ManxPower-work | ~ask |
| 14:56.25 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 14:56.57 | ManxPower-work | girlny: Are you also using a GUI for Asterisk? |
| 14:57.03 | [TK]D-Fender | dennis00: [usernamebudget] should have "insecure=port,invite" |
| 14:57.19 | girlny | im usinG freepbx 2.61 |
| 14:57.24 | ManxPower-work | ~freePBX |
| 14:57.25 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 14:57.39 | tzafrir_laptop | girlny, please pastebin the output of: cdr show status |
| 14:58.17 | p3nguin | I did it and there is no such command. |
| 14:58.17 | tzafrir_laptop | If you don't have cdr.conf, I believe cdr will be disabled altogether |
| 14:58.24 | *** join/#asterisk cjk (n=cjk@vodsl-10270.vo.lu) |
| 14:58.24 | ManxPower-work | tzafrir_laptop: Thank you for volunteering to help girlny with his/her FreePBX issues. |
| 14:58.38 | dennis00 | [TK]D-Fender: where? |
| 14:58.41 | tzafrir_laptop | Missing cdr.conf is not a freepbx issue |
| 14:58.46 | girlny | Verbosity is at least 3CLI> |
| 14:58.46 | girlny | domU-12-31-39-01-7D-16*CLI> cdr status |
| 14:58.46 | girlny | CDR logging: enabled16*CLI> |
| 14:58.46 | girlny | CDR mode: simple-7D-16*CLI> |
| 14:58.46 | girlny | CDR output unanswered calls: no |
| 14:58.47 | girlny | CDR registered backend: Adaptive ODBC |
| 14:58.48 | dennis00 | [TK]D-Fender: there are a lot of usernamebudgets. |
| 14:58.52 | *** join/#asterisk Arsenick (n=rpurcell@modemcable022.82-21-96.mc.videotron.ca) |
| 14:58.53 | tzafrir_laptop | ~pb |
| 14:58.54 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , or apt-get install pastebinit |
| 14:58.56 | p3nguin | girlny: PASTEBIN |
| 14:58.57 | ManxPower-work | tzafrir_laptop: looks to me like it is. |
| 14:59.32 | cjk | hi, my dtmfs work fine, except that i have to push # twice in order to get it recognised. my system is sip phone -> asterisk1 -> asterisk2 . the # dtmf arrives fine on asterisk1 according to the logs but not on asterisk2 |
| 14:59.36 | [TK]D-Fender | dennis00: in each |
| 14:59.49 | p3nguin | Just because someone uses FreePBX does not necessarily make the problem a FreePBX problem. |
| 14:59.54 | [TK]D-Fender | cjk: tT <- dial options |
| 14:59.58 | ManxPower-work | cjk: remove the w, W, t, T, k, and K options to your Dial line. |
| 15:00.10 | tzafrir_laptop | girlny, so I guess I was slightly wrong. The default is to have the CSV CDR enabled, I guess |
| 15:00.10 | *** part/#asterisk benngard (n=benngard@213.88.138.230) |
| 15:00.20 | ManxPower-work | p3nguin: it does when the next time the person presses the APPLY button on their GUI and get all their config files rewritten. |
| 15:00.25 | tzafrir_laptop | You're probably interested in the logging to mysql |
| 15:00.33 | tzafrir_laptop | Start with the sample cdr.conf |
| 15:00.49 | dennis00 | [TK]D-Fender: even in the register string? |
| 15:00.49 | cjk | [TK]D-Fender, ManxPower-work thanks, thats what i thought, any way to do this without removing the tT? |
| 15:01.07 | ManxPower-work | cjk: you have # configured to do DTMF based transfers. |
| 15:01.11 | [TK]D-Fender | dennis00: in the [31xxxxxx] peers dammit. |
| 15:01.12 | girlny | tzafrir_laptop how do i enable csv or mysql |
| 15:01.31 | [TK]D-Fender | cjk: It's F-ing HIJACKING them for transfers! What do you think? :p |
| 15:01.33 | tzafrir_laptop | hmmm.... my mistake. cdr_mysql.so is a completely independent module |
| 15:01.41 | [TK]D-Fender | cjk: So stop using DTMF transfers |
| 15:01.50 | *** join/#asterisk diatonic_afk (n=diatonic@mail.clearwater-research.com) |
| 15:01.57 | tzafrir_laptop | girlny, try: cdr mysql status |
| 15:02.15 | ManxPower-work | cjk: and spend the extra $2 and get a phone that has a transfer feature. |
| 15:02.48 | cjk | ok thanks for the help |
| 15:03.10 | cjk | even though i think asterisk should forward the #, even it its used in feautres.conf but i find a way to deal with iut |
| 15:03.17 | girlny | No such command 'cdr mysql status' (type 'help cdr mysql status' for other possible commands) |
| 15:04.05 | p3nguin | "asterisk should forward the #" <-- HUH?! |
| 15:04.08 | ManxPower-work | cjk: I've not used DTMF based transfers in like 5 years. For the most part only n00bs, cheapskates, and people with "special needs" (PBX needs, not other needs) use it. |
| 15:04.27 | dennis00 | [TK]D-Fender: http://pastebin.ca/1751026 |
| 15:04.31 | tzafrir_laptop | I guess something is wrong with the mysql logging module. What version of asterisk is it? |
| 15:04.46 | girlny | 1.605 |
| 15:05.01 | ManxPower-work | This is going to be funny. |
| 15:05.18 | girlny | the is no myslq module on my module forder |
| 15:05.57 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
| 15:06.09 | cjk | ManxPower-work, single number reachability where calls are forwarded to your mobile and you want to do transfers with your mobile |
| 15:06.32 | ManxPower-work | cjk: That is an example of one of the "special needs". 8-) |
| 15:06.33 | *** join/#asterisk mrbnet (n=mrbnet@74-95-100-233-Minnesota.hfc.comcastbusiness.net) |
| 15:07.07 | ManxPower-work | cjk: make those forwarded calls use a Dial with the T option and all other calls not use that option. |
| 15:07.11 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 15:07.29 | ManxPower-work | sorry, that should have been the "t" option, not "T" |
| 15:07.34 | [TK]D-Fender | dennis00: what is [311]? This is NOT a normal budgetphone DID authing peer |
| 15:07.48 | dennis00 | 311 is my phone number, I changed it. |
| 15:07.52 | dennis00 | from the actual long number. |
| 15:07.56 | cjk | yeah i see, i should optimise |
| 15:07.57 | dennis00 | it is also my username |
| 15:08.02 | ManxPower-work | dennis00: don't change info except for passwords |
| 15:08.06 | dennis00 | ok. |
| 15:08.07 | tzafrir_laptop | girlny, you seem to have a problem with the module cdr_mysql.so . It comes from asterisk-addons (up until Asterisk 1.6.2) |
| 15:08.08 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 15:08.20 | [TK]D-Fender | dennis00: ....where's the new calla ttempt? |
| 15:08.22 | tzafrir_laptop | girlny, How was asterisk installed? |
| 15:08.39 | dennis00 | [TK]D-Fender: Do you want a pastebin from asterisk? |
| 15:09.18 | [TK]D-Fender | .... |
| 15:09.55 | dennis00 | [TK]D-Fender: I am so sorry, but I don't get it. |
| 15:10.06 | p3nguin | Nah, a pastebin from the refrigerator would be more useful. |
| 15:10.07 | dennis00 | I tried another call after adding insecure=very. |
| 15:10.14 | [TK]D-Fender | dennis00: PLACE ANOTHER CALL TO TEST IT |
| 15:10.28 | dennis00 | I have, it still fails. |
| 15:10.30 | ManxPower-work | Does anyone happen to know the sound file played to indicate someone left a MeetMe? |
| 15:10.47 | ManxPower-work | dennis00: He's not looking for success, he's looking for debug info of a failure. |
| 15:10.49 | girlny | I used a ec2 ami, that has freepbx and asterisk everything works perfect but the cdrs |
| 15:11.03 | dennis00 | Ok, here is the pastebin. http://pastebin.ca/1751030 |
| 15:11.23 | Dovid | <PROTECTED> |
| 15:11.32 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:12.01 | Dovid | Morning TK: |
| 15:12.06 | [TK]D-Fender | dennis00: full configs masking ONLY passwords, and make sure you apply your changes |
| 15:13.39 | *** join/#asterisk n3hxs (n=HAMming@75-145-10-45-Delmarva.hfc.comcastbusiness.net) |
| 15:13.52 | tzafrir_laptop | girlny, what's the output of: module show like mysql ? |
| 15:14.12 | *** join/#asterisk DarkFibre (n=dmelouk@127.159.119.70.cfl.res.rr.com) |
| 15:14.33 | *** join/#asterisk skymeyer (n=skymeyer@mailout.itconnect.be) |
| 15:14.45 | *** join/#asterisk angryuser_ (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) |
| 15:14.49 | *** join/#asterisk mrchrisadams (n=mrchrisa@87-194-125-43.bethere.co.uk) |
| 15:15.00 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 15:15.03 | dennis00 | http://pastebin.ca/1751038 |
| 15:15.37 | *** join/#asterisk grnmtn (n=grnmtn@65.183.149.153) |
| 15:15.59 | p3nguin | # |
| 15:16.05 | p3nguin | register => 31107142866@sip1.budgetphone.nl:bpassword:31107142866@sip1.budgetphone.nl/31107142866 |
| 15:16.08 | p3nguin | Is that right? |
| 15:16.15 | dennis00 | Well, it works for outgoing calls. |
| 15:16.16 | *** join/#asterisk Mark22 (n=mark@unaffiliated/mark21) |
| 15:16.21 | dennis00 | And it is right according to budgetphone Yes. |
| 15:16.27 | p3nguin | Seems screwy to me. |
| 15:16.43 | [TK]D-Fender | ....... |
| 15:16.51 | [TK]D-Fender | REGISTER has NOTHING to do with outgoing calls |
| 15:16.59 | [TK]D-Fender | and nothing to do with AUTHING incoming calls |
| 15:17.10 | ManxPower-work | registration has nothing to do with outgoing calls |
| 15:17.47 | girlny | there are 175 modules |
| 15:18.01 | *** join/#asterisk dennis00 (i=dennis00@unaffiliated/dennis00) |
| 15:18.05 | dennis00 | So, that went wrong. |
| 15:19.01 | dennis00 | Hmm,, I made a change which sends all calls to my phone now. |
| 15:19.46 | *** part/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 15:20.41 | dennis00 | when I comment out inbound, which is before outbound, I can send outgoing calls again |
| 15:21.41 | dennis00 | Ahhhh, I had it working. |
| 15:22.16 | dennis00 | I guess I didn't. I can call from Asterisk->Asterisk, but not Voipbuster->Asterisk (budgetphone) |
| 15:22.31 | *** part/#asterisk Mark22 (n=mark@unaffiliated/mark21) |
| 15:22.46 | dennis00 | [TK]D-Fender: do you have any suggestoins? |
| 15:22.57 | *** join/#asterisk t (i=tom@freenode/staff/tomaw) |
| 15:23.52 | *** join/#asterisk jdnWEST (n=jdn@mx1.westparkcom.net) |
| 15:23.54 | *** join/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 15:24.11 | p3nguin | Calls to and from voipbuster should be easy to get going -- it's this budgetphone thing I am not familiar with. |
| 15:24.44 | jdnWEST | If i'm using a cisco router to register with my voip provider, and just dial peer the traffic to an asterisk box (no transcoding) will I need any DSPS? |
| 15:24.51 | [TK]D-Fender | moves on to more productive matters |
| 15:25.28 | [TK]D-Fender | jdnWEST: why is CISCO involved with the call at all? |
| 15:25.55 | [TK]D-Fender | jdnWEST: And * doesn't require hardware to process those calls |
| 15:26.12 | [TK]D-Fender | (more thant he PC its running on) |
| 15:26.18 | jdnWEST | My sip provider can only handle 2 end points. the legacy pbx I have only takes PRIs (2 pri's in a card = 8k) |
| 15:27.02 | jdnWEST | I was going to have my 2 cisco routers act as end points and load balence it across multiple asterisk boxes with 2PRI's that would then send the calls to the legacy pbx |
| 15:27.34 | dennis00 | p3nguin: I see. But my VOIP-in from Voipbuster did not work anyhow. |
| 15:27.47 | p3nguin | I'm sure it would have if you had configured it correctly. |
| 15:27.53 | [TK]D-Fender | jdnWEST: Please draw a clearer picture |
| 15:27.57 | dennis00 | I used X-lite to connect to their SIP. |
| 15:27.58 | *** join/#asterisk lordmortis (n=lordmort@203-206-67-161.dyn.iinet.net.au) |
| 15:28.07 | jdnWEST | I'm trying to setup trunking into my legacy pbx. |
| 15:28.15 | jdnWEST | Legacy pbx will only take PRI's |
| 15:28.42 | [TK]D-Fender | jdnWEST: So shove a PRI card into your * and be done with it |
| 15:28.54 | *** join/#asterisk melter (n=Melter@dyn065.morrill-116.ndsu.NoDak.edu) |
| 15:29.11 | jdnWEST | I need 8 PRI's. |
| 15:29.23 | jdnWEST | If I do 4 and 4 i'll loose 1/2 my trunks doing maintence. |
| 15:29.26 | [TK]D-Fender | jdnWEST: You just said 2 earlier |
| 15:29.35 | jdnWEST | 2 per device is what I want to do. |
| 15:29.46 | girlny | were can i find affordable paid support for freepbx and asterisk |
| 15:29.47 | jdnWEST | because my legaxy pbx's T1 cards are 8k each. |
| 15:30.01 | jdnWEST | and I only want to have to have 1 extra card in it. |
| 15:30.07 | melter | is it possible to use asterisk to run a 1-800 or 1-900 number that would play pre-recorded audio? |
| 15:30.13 | dennis00 | p3nguin: Then can help me to get my monkeys working? |
| 15:30.21 | [TK]D-Fender | melter: Yes |
| 15:30.23 | jdnWEST | melter -> Just point it at a vmail box |
| 15:30.31 | [TK]D-Fender | ..... |
| 15:30.34 | [TK]D-Fender | NO V-MAIL. |
| 15:30.34 | dennis00 | when I dial 101, it should say => exten => 101,n,Playback(tt-monkeys) |
| 15:30.36 | [TK]D-Fender | gah |
| 15:30.50 | p3nguin | dennis00: Where's priority 1? |
| 15:31.05 | p3nguin | exten => 101,1,Playback() |
| 15:31.14 | dennis00 | exten => _X.,1,NoOp(*** Incoming Call! *** ) |
| 15:31.21 | melter | what kind of card would i need? i think 1 or 2 phone lines would be enough |
| 15:31.25 | dennis00 | No) exten => 101,1,Answer() |
| 15:31.36 | jdnWEST | melter, no need for a card. Just get a VOIP number. |
| 15:31.37 | dennis00 | That should work under inbound right? |
| 15:31.37 | *** join/#asterisk tgunr (n=tgunr@cust-66-249-166-12.static.o1.com) |
| 15:31.43 | dennis00 | exten => 101,1,Answer() exten => 101,n,Playback(tt-monkeys) exten => 101,n,Hangup() |
| 15:31.50 | melter | jdnWEST: thanks! |
| 15:31.56 | [TK]D-Fender | [10:31]<jdnWEST>melter, no need for a card. Just get a VOIP num <- don't assume HOW they want to get their calls |
| 15:32.05 | p3nguin | dennis00: That looks like 101 will work. |
| 15:32.09 | [TK]D-Fender | melter: HOW you want to get it is up to you |
| 15:32.19 | dennis00 | p3nguin: but must I put it under inbound or outbound? |
| 15:32.25 | [TK]D-Fender | melter: And if that ends up being a line you're paying for already then hardware is in order |
| 15:32.38 | p3nguin | dennis00: Put it whereever you want it. Just make sure your peer's context points to it. |
| 15:33.30 | melter | [TK]D-Fender: i'd thought about buying a cheap, dedicated linux box, should be <$500 |
| 15:33.49 | [TK]D-Fender | melter: Still has nothing to do with HOW you want the call to arrive at your server |
| 15:34.13 | [TK]D-Fender | melter: what tech, how many simultaneous calls to support, etc |
| 15:34.41 | p3nguin | melter: My toll-free number arrives at my box over IAX2, but I could change it to SIP if I want. The only "card" I need an a network interface CARD. |
| 15:34.41 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 15:34.56 | melter | [TK]D-Fender: all i want to do is have someone dial either a 1-800 or 1-900 number (haven't decided), pick something from a menu, and play a pre-recorded message |
| 15:35.20 | melter | [TK]D-Fender: 1 call at a time is probably ok, but 2 would be nice |
| 15:35.28 | [TK]D-Fender | melter: HOW AMNY CALLS AT AT TIME? |
| 15:35.29 | p3nguin | Easy. BackGround() and Playback() |
| 15:35.44 | melter | [TK]D-Fender: real low volume, and only in the summer (it's for weather-related messages) |
| 15:36.03 | [TK]D-Fender | melter: Well * can process calls over whatever you set up with it. Look at what services will get you what you need. |
| 15:36.16 | melter | [TK]D-Fender: 2 at a time would enough |
| 15:36.48 | jdnWEST | melter: Your welcome to use asterisk, but you could just get a voicemail account and use that and not have to manage hardware. |
| 15:37.06 | p3nguin | melter: You can get a free regular DID (not toll free, not 900) from many ITSPs. |
| 15:37.18 | p3nguin | melter: They will deliver calls to your * server with SIP or IAX2. |
| 15:37.20 | [TK]D-Fender | melter: Do you already have these lines? Have you looked at what this service will cost? |
| 15:37.56 | melter | [TK]D-Fender: no, and no, i'm just starting to look into this |
| 15:38.03 | [TK]D-Fender | ~itsplist-us |
| 15:38.04 | infobot | well, itsplist-us is Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://www.voicepulse.com/connect/ , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net |
| 15:38.05 | [TK]D-Fender | ^^^^^^ |
| 15:38.07 | dennis00 | <PROTECTED> |
| 15:38.16 | melter | i'd have 72 different audio messages, and each would update about every 10 minutes |
| 15:38.17 | [TK]D-Fender | metget shopping |
| 15:38.22 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:38.30 | p3nguin | melter: sipgate, ipkall, and ipcomms will all give you FREE DIDs if you just sign up and ask. |
| 15:38.32 | [TK]D-Fender | melter: get shopping |
| 15:38.44 | [TK]D-Fender | ....... |
| 15:38.48 | p3nguin | For toll-free, check VoIP.ms |
| 15:38.51 | [TK]D-Fender | dear lord nobody reads. |
| 15:38.53 | [TK]D-Fender | ever |
| 15:39.12 | jdnWEST | fender: why would we do that :P |
| 15:39.15 | ManxPower-work | [TK]D-Fender: I don't use MeetMe much, do you recall if there is a meep-meep sound when someone leaves a conference? |
| 15:39.16 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 15:39.31 | [TK]D-Fender | ManxPower-work: You porbably used it more then I |
| 15:39.43 | ManxPower-work | [TK]D-Fender: I've not used it in like 2 years. 8-| |
| 15:39.58 | [TK]D-Fender | ManxPower-work: I've probably not used ti more than 2 times :p |
| 15:40.12 | ManxPower-work | [TK]D-Fender: You also hate conference calls? |
| 15:42.03 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 15:42.40 | areay | since my server restart, it seems cdr_mysql has stopped working... it was working fine before, but now i'm getting "column count doesn't match value count at row 1" in my CLI every time asterisk tries to add a row to the database... |
| 15:43.16 | ManxPower-work | areay: sounds like you upgraded Asterisk |
| 15:43.36 | areay | ManxPower-work, not yet... i'm ready to (i think), but i wanted to make sure everything was working ok first |
| 15:43.43 | p3nguin | MeetMe has join/leave sounds. |
| 15:43.54 | ManxPower-work | p3nguin: thanks. |
| 15:44.22 | p3nguin | The q option silences them, though. |
| 15:44.38 | ManxPower-work | p3nguin: *nod* I'm trying to play the exact same sound in the dialplan |
| 15:44.58 | areay | ManxPower-work, is there a way of dumping the specific query that asterisk has sent? maybe it will tell me more... |
| 15:45.08 | ManxPower-work | areay: I don't know |
| 15:45.12 | dennis00 | my Asterisk currently hangs up straight, what could this be? http://pastebin.ca/1751071 |
| 15:46.13 | kaldemar | dennis00: anything. show what happens before the BYE message. |
| 15:46.13 | *** join/#asterisk The_Boy_Wonder (n=vossel@asterisk/batman-developer/dvossel) |
| 15:46.42 | muiro | I need someone to clear something up about Realtime for me. When using realtime, when I set up the database table, do the fields I need in the table have a 1 to 1 correlation with the options available in the .conf file that would be used for that module when using a static configuration? |
| 15:47.10 | dennis00 | http://pastebin.ca/1751074 |
| 15:47.32 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 15:49.14 | dennis00 | I did change my extensions.conf and sip.conf, that might be it. http://pastebin.ca/1751077 |
| 15:49.49 | p3nguin | manxpower-work: The sound when a caller leaves is quite unusual. I don't know how I would duplicate it. |
| 15:51.09 | p3nguin | dennis00: Every place you have a ,r in a Dial() command, delete it. |
| 15:51.45 | p3nguin | Is there an infobot entry for Dial()'s "r" and "R"? |
| 15:52.17 | dennis00 | p3nguin: I changed it, still same issue. |
| 15:52.25 | p3nguin | What's the problem now? |
| 15:52.34 | dennis00 | It hangs up within a second. |
| 15:52.40 | dennis00 | I cannot send outgoing calls anymore. |
| 15:53.06 | p3nguin | To what provider? |
| 15:53.15 | dennis00 | Ohh god. I am being stupid again. |
| 15:53.25 | dennis00 | I had remember a wwrong number :( |
| 15:53.33 | dennis00 | I thought it would at least say that in a voice. |
| 15:53.38 | dennis00 | Calling out does work. |
| 15:54.05 | ManxPower-work | ~r |
| 15:54.06 | infobot | extra, extra, read all about it, r is The "r" option to Dial will override any sounds you should be hearing and provide a fake ringing sound to the caller. You generally want the caller to hear the sounds they are supposed to hear, not a fake ringing sound. The caller will hear ringing without the "r" option. Using the "r" option is an edge case and should not normally be used or needed. |
| 15:55.02 | dennis00 | I see. OK, I removed it. |
| 15:55.11 | *** join/#asterisk Beers218 (i=45c6b055@gateway/web/freenode/x-omjulmgigoowczpx) |
| 15:56.38 | dennis00 | Now it says calling and 'busy here' when I try to call my own number via voipbuster->budgetphone |
| 15:57.43 | dennis00 | I seriously need a program that allows multiple SIP accounts for testing. |
| 15:58.07 | dennis00 | Now it says 'person for extension 1001 is unavailable' when I call my own number, which is a good thing. |
| 15:58.15 | dennis00 | As 1001 is an extension on my server. |
| 15:59.58 | areay | ManxPower-work, turns out restarting asterisk fixed it lol... maybe i do need to upgrade |
| 16:00.08 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 16:00.36 | *** join/#asterisk blaxthos (n=mcp@69.61.106.93) |
| 16:00.40 | *** part/#asterisk blaxthos (n=mcp@69.61.106.93) |
| 16:00.42 | dennis00 | exten => s < what is the s? |
| 16:01.23 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
| 16:03.15 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 16:03.15 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 16:03.29 | kaldemar | dennis00: http://www.voip-info.org/wiki/view/Asterisk+standard+extensions |
| 16:04.07 | dennis00 | thank you! |
| 16:04.45 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
| 16:05.46 | *** join/#asterisk McLazarus (n=McLazaru@dogpile.mcallister.ws) |
| 16:05.47 | *** join/#asterisk c0rnoTa (n=c0rnoTa@188.162.29.116) |
| 16:06.04 | *** part/#asterisk c0rnoTa (n=c0rnoTa@188.162.29.116) |
| 16:06.57 | bmoraca_work | dennis00, did you read the book? |
| 16:07.12 | dennis00 | bmoraca_work: I have a book yes. |
| 16:07.32 | bmoraca_work | did you READ THE book? |
| 16:07.34 | bmoraca_work | as in... |
| 16:07.36 | bmoraca_work | ~book |
| 16:07.37 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:07.42 | dennis00 | I read parts of it. |
| 16:08.00 | TheDavidFactor | is there a way to tell what IP address asterisk is running on from inside the dialplan? |
| 16:08.01 | bmoraca_work | just making sure...it's got much of the info you'll need |
| 16:08.16 | bmoraca_work | TheDavidFactor, why would you need to? |
| 16:08.20 | kaldemar | TheDavidFactor: sure. |
| 16:08.30 | p3nguin | dennis00: s usually matches when no phone number is included. |
| 16:08.45 | dennis00 | p3nguin: in sip.conf? |
| 16:08.49 | *** join/#asterisk Alagar (n=Administ@122.164.41.12) |
| 16:08.50 | p3nguin | no |
| 16:08.54 | p3nguin | the 's' extension |
| 16:08.57 | ManxPower-work | I hate Verizon. They fixed a POTS line, but turned it from Groundstart to Loopstart, so it stopped working. |
| 16:09.20 | TheDavidFactor | kademar, anything more then "sure"? :-) |
| 16:09.22 | beek | ManxPower-work: and let me guess, you had to find that out on your own. |
| 16:09.32 | dennis00 | If you look at inboud, you see it should always work? http://pastebin.ca/1751097 |
| 16:09.40 | dennis00 | It's so simple. |
| 16:10.17 | ManxPower-work | beek: someone else found it, it was not my ticket to work. |
| 16:11.01 | girlny | when i do module show like mysql shows 0 modules |
| 16:11.08 | kaldemar | TheDavidFactor: for example with function SHELL |
| 16:11.14 | *** join/#asterisk titter (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 16:11.21 | TheDavidFactor | thanks |
| 16:11.21 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:80a:1c67:a077:acde) |
| 16:11.30 | cusco | is there a sip header equivalent for iax2? |
| 16:11.45 | ManxPower-work | cusco: your question makes no sense |
| 16:11.49 | cusco | :( |
| 16:11.57 | ManxPower-work | Yes, IAX2 has headers, but not like SIP. |
| 16:11.58 | [TK]D-Fender | cusco: No. IAX2 is not a routed protocol |
| 16:11.59 | kaldemar | cusco: IAX2 has headers too, so yes. |
| 16:12.03 | [TK]D-Fender | (not so much) |
| 16:12.44 | ManxPower-work | cusco: Do you mean "How can I ACCESS IAX2 headers like I can access them on a SIP channel?" |
| 16:12.46 | cusco | I was passing the UNIQUEID trough sip header |
| 16:12.50 | tzafrir_laptop | girlny, you likely don't have asterisk-addons installed |
| 16:12.53 | cusco | but now we are connecting trough uax2 |
| 16:12.54 | cusco | iax2 |
| 16:13.00 | tzafrir_laptop | grab it and build it, I guess |
| 16:13.16 | ManxPower-work | cusco: vague questions usually get wrong answers -- as in your case. |
| 16:13.24 | *** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca) |
| 16:13.40 | kaldemar | cusco: do you mean like an X-header? |
| 16:13.47 | ManxPower-work | cusco: Have you read channelvars.tex? look for the _ and __ (underscore and doubleunderscore) stuff. |
| 16:14.17 | Beers218 | anyone know if its possible to configure asterisk to use the intercom when doing a attended transfer rather than just ringing the destination extension? |
| 16:14.21 | cusco | kaldemar: yes |
| 16:14.34 | cusco | ManxPower-work: i understand |
| 16:14.42 | kaldemar | cusco: there's func IAXVAR for similar functionality. |
| 16:15.23 | cusco | thanks kaldemar |
| 16:15.46 | ManxPower-work | Beers218: do you understand how "intercom" works? |
| 16:15.58 | tzafrir_laptop | girlny, does that EC2 instance of yours include a source installation of Asterisk as well? |
| 16:16.08 | [TK]D-Fender | Beers218: Head back to #freepbx <- |
| 16:16.28 | [TK]D-Fender | Beers218: A transfer is a transfer. Where you send it is up to you and what it does is up to you |
| 16:16.31 | ManxPower-work | [TK]D-Fender: thanks for the intercept. I was going to help him. |
| 16:16.40 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 16:16.41 | Beers218 | ManxPower-work: the way i understand it is that the phones are configured to autoanswer |
| 16:16.49 | ManxPower-work | Not that my help will do any good to a FreePBX user. |
| 16:17.57 | ManxPower-work | Beers218: you can ask here all day, but none of the answers will apply to you if you are running FreePBX |
| 16:18.05 | [TK]D-Fender | Beers218: the EXTEnsION you dial chooses what it does, and an ATTENDED TRANSFER isn't a transfer until it is completed, it is a NEW CALL |
| 16:18.42 | Beers218 | yes, i know all this, i just wanted to know, in general, if it is possible |
| 16:18.51 | girlny | tzafrir_laptop yes it does include the source |
| 16:19.08 | ManxPower-work | If someone transfered a call to me that was auto-answered, my next task would to take my hardcover Advanced Programming in the Unix Enviroment and beat the user over the head. |
| 16:19.25 | ManxPower-work | Beers218: yes, in theory, it is possible. |
| 16:19.29 | girlny | tzafrir_laptop yes it includes the source |
| 16:19.53 | tzafrir_laptop | girlny, so grab asterisk-addons for 1.6.0 from http://downloads.asterisk.org/pub/telephony/asterisk/ |
| 16:19.57 | tzafrir_laptop | and build it |
| 16:20.29 | Beers218 | basically my boss wants an attended transfer to page the person so you can talk to him/her and say |
| 16:20.40 | Beers218 | hey you got a call coming, hang up and it rings on the destination extension. |
| 16:20.45 | tzafrir_laptop | http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-addons-1.6.0-current.tar.gz |
| 16:21.15 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 16:21.31 | tzafrir_laptop | girlny, I guess you'll need to install mysql-devel or whatever in order for the mysql stuff to build |
| 16:23.09 | [TK]D-Fender | Beers218: No, that order is not really doable |
| 16:23.54 | ManxPower-work | Beers218: but if you can live without RINGING the extension that should be do-able |
| 16:24.01 | dennis00 | [TK]D-Fender: should this not always dial extension 1001 for unknown numbers? exten = s,1,Dial(SIP/1001) |
| 16:24.10 | ManxPower-work | dennis00: no. |
| 16:24.11 | [TK]D-Fender | dennis00: NO. |
| 16:24.25 | [TK]D-Fender | dennis00: "s" is not a mircale catch-all |
| 16:24.25 | ManxPower-work | That would always dial the extension 1001 when the number was NOT SENT. |
| 16:24.25 | [TK]D-Fender | ~stdextens |
| 16:24.26 | infobot | [~stdextens] The "s" Standard Extension : Where a call goes to when * does not know the destination of the call. Ex : Calls coming in on FXO ports (no DID), or from an ITSP that doesn't specify or where it was not set in the REGISTER line, or FXS port goes off-hook and "immediate=yes" in zapata.conf. "s" is also used to make IVRs & macros. |
| 16:24.45 | dennis00 | Well, my VOIPBuster calls anonymously. |
| 16:24.57 | [TK]D-Fender | dennis00: No, they target an EXTENSION |
| 16:25.07 | Alagar | happy pongal |
| 16:25.48 | dennis00 | exten = _X,1,Dial(SIP/1001) or exten = actualnumber,1,Dial(SIP/1001) is not really working either. |
| 16:26.03 | [TK]D-Fender | dennis00: because that will match only a SINGLE DIGIT |
| 16:26.09 | Alagar | happy pongal |
| 16:26.15 | [TK]D-Fender | dennis00: You dont' seem to have understood dialplan patterns at all |
| 16:26.20 | Beers218 | ManxPower-work: what do you mean by "without ringing the extension" im confused there. |
| 16:26.25 | ManxPower-work | I think maybe dennis00 needs to reread dialplan patterns. |
| 16:26.26 | dennis00 | I had made a list of 6, for all options. But that failed too. |
| 16:26.29 | [TK]D-Fender | dennis00: Go read chapter 5 of THE BOOK again |
| 16:26.31 | [TK]D-Fender | ~book |
| 16:26.32 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 16:26.47 | dennis00 | it's still downloading |
| 16:26.54 | [TK]D-Fender | Beers218: there is no "ring them AFTER you hang up" |
| 16:26.58 | ManxPower-work | Beers218: You said "(11:20:29 AM) Beers218: basically my boss wants an attended transfer to page the person so you can talk to him/her and say hey you got a call coming, hang up and it *******rings********* on the destination extension." |
| 16:26.59 | [TK]D-Fender | beethat is not happening |
| 16:27.32 | *** join/#asterisk Heretic (n=fallen@dsl-246-68-96.telkomadsl.co.za) |
| 16:27.51 | Beers218 | ok, thats what i thought. i just wanted to get a professional opinion |
| 16:28.21 | ManxPower-work | Attended transfers don't complete with RINGING the destination again. They complete when the person doing the transfer completes the transfer. There is no RINGING involved. |
| 16:30.36 | *** part/#asterisk ibolmo (n=ibolmo@cpe-67-11-89-134.rgv.res.rr.com) |
| 16:30.50 | p3nguin | dennis00: The last pastebin of extensions should work, but since you haven't shown the sip configuration after you've mangled it nor the debug, we can't help you any better. |
| 16:31.33 | dennis00 | p3nguin: I am confused a lot, give me a second to recover and I'll post both files. |
| 16:32.04 | ruyo | What can it mean when a compilation gets stuck (20+ mins) on "Entering directory"? |
| 16:32.11 | p3nguin | dennis00: If you'll stop guessing and start paying better attention, that could be in your best interest. |
| 16:32.34 | ManxPower-work | p3nguin is growing up to be a cynical BOFH like the rest of us! |
| 16:32.40 | *** join/#asterisk verywiseman (n=khaled@unaffiliated/verywiseman) |
| 16:32.50 | p3nguin | I have my moments. |
| 16:33.17 | Beers218 | ManxPower-work: right, i figured that. |
| 16:33.58 | ManxPower-work | p3nguin: Stick around here for a few years and you'll be just like me and [TK]D-Fender |
| 16:34.41 | [TK]D-Fender | p3nguin: I'm patented.. you won't live long enough to get the rights ;) |
| 16:34.45 | dennis00 | p3nguin: Sure, I will, but I honestly thought my config was correct. I will post it. |
| 16:34.53 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 16:35.42 | dennis00 | http://pastebin.ca/1751127 |
| 16:37.51 | ManxPower-work | dennis00: If you have context=da-isdn in sip.conf then YOU MUST have [da-isdn] in your dialplan extensions.conf |
| 16:38.11 | dennis00 | ManxPower-work: Voipbuster does not have to work, but budgetphone. |
| 16:38.11 | ManxPower-work | This is always the case. |
| 16:38.37 | ManxPower-work | dennis00: I would not expect voipbuster to work. You are sending calls to the da-isdn context which does not exist. |
| 16:38.48 | verywiseman | i am working on * 1.6 , and i install asterisk-addons , to store cdr to mysql, and i followed steps in "http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql", but when i run "asterisk -cvvv" , this Warning msg appear: WARNING[31857]: config.c:1111 process_text_line: parse error: No category context for line 14 of /etc/asterisk/cdr_mysql.conf , and this is cdr_mysql.conf file "http://fpaste.org/6tCx/" |
| 16:39.01 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 16:39.02 | girlny | tzafrir_laptop intall againg the add onds and still shos 0 in module show like mysql |
| 16:39.04 | casix | hello |
| 16:39.25 | ManxPower-work | verywiseman: Do you really want ;[global] |
| 16:40.01 | minotaur01 | is there a way to force asterisk to use a specific ip route? |
| 16:40.04 | ManxPower-work | Maybe you want "[global]" instead of "[global]" |
| 16:40.08 | ManxPower-work | minotaur01: no. |
| 16:40.15 | ManxPower-work | minotaur01: that is up to your network. |
| 16:40.34 | tzafrir_laptop | girlny, try: module load cdr_addon_mysql.so |
| 16:40.41 | tzafrir_laptop | (or just restart asterisk) |
| 16:40.41 | dennis00 | ManxPower-work: I changed it, but I want incoming calls to work and they fail miserably. |
| 16:40.56 | ManxPower-work | dennis00: you changed what from what to what? |
| 16:40.56 | verywiseman | ManxPower-work, thank you , i remove ; before [global] , it is working now :) |
| 16:41.02 | casix | I have some many errors like this "chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 20" in an asterisk 1.4.21.2 with asterisk-addons-1.4.7, zaptel-1.4.12.1, libpri-1.4.7 . Is this a hardware problem? |
| 16:41.08 | dennis00 | ManxPower-work: I added [da-isdn] in extensions.conf |
| 16:41.23 | ManxPower-work | dennis00: and did you put some exten => line to match an incoming call in that context? |
| 16:42.06 | ManxPower-work | dennis00: now re-pastebin the configs |
| 16:42.10 | dennis00 | I do not have voipin for voipbuster |
| 16:42.12 | [TK]D-Fender | dennis00: You seem to be asking us how much it will cost to fix your car... based on a picture of how it looked BEFORE YOU CRASHED IT. |
| 16:42.19 | tzafrir_laptop | casix, what hardware is it? |
| 16:42.22 | Katty | tries to not be cranky and breathe fire at people. |
| 16:42.24 | [TK]D-Fender | dennis00: Show us the &#^$ING CALL |
| 16:42.29 | dennis00 | I only changed 1 thing. |
| 16:42.36 | girlny | tzafrir_laptop how do i try module load cdr_addon_mysql.so |
| 16:42.40 | [TK]D-Fender | Katty: I'll be happy to vent for you :p |
| 16:42.48 | casix | tzafrir_laptop: Digium, Inc. Wildcard TE410P Quad-Span |
| 16:43.04 | minotaur01 | ManxPower-work: then how come the sip trafic is not going through my VPN and if i do a trace route it uses the VPN |
| 16:43.20 | girlny | tzafrir_laptop [root@domU-12-31-39-01-7D-16 ~]# module load cdr_addon_mysql.so |
| 16:43.21 | girlny | -bash: module: command not found |
| 16:43.35 | *** join/#asterisk coppice (n=chatzill@106.202.17.210.dyn.pacific.net.hk) |
| 16:43.40 | [TK]D-Fender | girlny: ASTERISK CLI |
| 16:43.44 | tzafrir_laptop | girlny, in the asterisk CLI |
| 16:43.46 | [TK]D-Fender | girlny: please get with the program |
| 16:43.51 | girlny | ok thanks |
| 16:44.04 | [TK]D-Fender | girlny: They are being dispensed in a tray to your left |
| 16:44.08 | ManxPower-work | minotaur01: I don't know, but Asterisk is not forcing any network routes or interfaces unless you use the very bad option bindaddr |
| 16:44.29 | dennis00 | http://pastebin.ca/1751143 |
| 16:44.38 | p3nguin | dennis00: Do you comprehend how contexts and dialplans work? |
| 16:44.39 | ManxPower-work | and with bindaddr it will only force the source IP, not a route |
| 16:44.49 | dennis00 | p3nguin: I thought I did. |
| 16:44.59 | dennis00 | brb, 1min |
| 16:45.04 | tzafrir_laptop | casix, is span 1 in alarm? |
| 16:45.06 | p3nguin | dennis00: What is the context for the incoming calls for your sip peer? |
| 16:45.07 | *** join/#asterisk uluatu (n=uluatu@200.146.5.244.dynamic.dialup.gvt.net.br) |
| 16:45.17 | [TK]D-Fender | dennis00: SIP/2.0 401 Unauthorized <- seems to say it... |
| 16:45.19 | minotaur01 | ManxPower-work: well its got me stumped |
| 16:45.25 | girlny | domU-12-31-39-01-7D-16*CLI> module load cdr_addon_mysql.so |
| 16:45.26 | girlny | Unable to load module cdr_addon_mysql.so |
| 16:45.26 | girlny | Command 'module load cdr_addon_mysql.so' failed. |
| 16:45.26 | p3nguin | dennis00: It says # |
| 16:45.27 | ManxPower-work | dennis00: OK, so a call comes in, matches [voipbuster] the call then gets sent to the da-isdn context in extensions.conf and finds no extensions to match the call to, then fails. |
| 16:45.29 | casix | tzafrir_laptop: no, there are no alarms |
| 16:45.43 | p3nguin | dennis00: It says context=da-isdn |
| 16:45.45 | dennis00 | it's a budgetphone number, not voipbuster |
| 16:45.51 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:46.11 | tzafrir_laptop | casix, can you pastebin the output of 'cat proc/zaptel/*' ? |
| 16:46.12 | p3nguin | dennis00: Are you dealing with the voipbuster peer or the budgetphone peer? |
| 16:46.36 | ManxPower-work | dennis00: turn on sip debug, pastebin the output of a failed call. No, do not use one provide to call the other provider. use a phone not linked to Astrisk |
| 16:47.25 | p3nguin | Really, we don't even need any debug... we just need dennis00 to understand how the peers work and what the context value means. |
| 16:47.31 | girlny | tzafrir_laptop Unable to load module cdr_addon_mysql.so |
| 16:47.39 | casix | tzafrir_laptop: http://pastebin.com/m579accb2 |
| 16:47.40 | p3nguin | Once he understands that and configures it correctly, it will work. |
| 16:48.28 | p3nguin | And we need dennis00 to pay better attention and answer questions fully. |
| 16:48.37 | tzafrir_laptop | girlny, and specific message before that? |
| 16:49.00 | minotaur01 | dose anyone here have a tutorial on how to install remote packet capture? |
| 16:49.16 | p3nguin | And we need dennis00 to stop dicking around with the configs until he understands condition 1. |
| 16:49.25 | p3nguin | s/understands/meets/ |
| 16:49.29 | tzafrir_laptop | minotaur01, install tcpdump from your distro |
| 16:49.53 | minotaur01 | will that let me use it with wireshark? |
| 16:50.19 | p3nguin | wireshark has some CLI tools, but why do you need wireshark when you have tcpdump? |
| 16:51.03 | ManxPower-work | minotaur01: use what with wireshark? |
| 16:51.04 | tzafrir_laptop | casix, may be harmless, but might be a symptom to something else: are channels 48-62 left unused intentionally? |
| 16:51.44 | girlny | tzafrir_laptop domU-12-31-39-01-7D-16*CLI> module load cdr_addon_mysql.so |
| 16:51.44 | girlny | Unable to load module cdr_addon_mysql.so |
| 16:51.44 | girlny | Command 'module load cdr_addon_mysql.so' failed. |
| 16:52.07 | minotaur01 | ManxPower-work: if i install tcpdump will it work with wireshark... i want to do remote packet captures with wireshark... |
| 16:52.24 | muiro | If I'm using MeetMeAdmin to kick one user or mute one user, how wil the users be named? |
| 16:52.49 | minotaur01 | p3nguin: wireshark has more tools then just packet capturing |
| 16:52.51 | ManxPower-work | minotaur01: I have no idea what you mean by "remote captures" |
| 16:52.59 | tzafrir_laptop | minotaur01, remote: from where? |
| 16:53.09 | ManxPower-work | You capture on a local interface. |
| 16:53.19 | minotaur01 | no |
| 16:53.42 | minotaur01 | i want to capture packets on my asterisk box from my pc |
| 16:53.45 | *** join/#asterisk VaGoNeTaS (n=cchavez@200.111.138.170) |
| 16:53.45 | p3nguin | Once you login on that remote device, it is no longer remote. |
| 16:53.47 | VaGoNeTaS | guys |
| 16:53.48 | ManxPower-work | minotaur01: That will not happen. |
| 16:53.51 | VaGoNeTaS | hello everybody |
| 16:54.00 | *** join/#asterisk Caplain (i=shayne@84-141.35-65.tampabay.res.rr.com) |
| 16:54.01 | VaGoNeTaS | i got a little question |
| 16:54.10 | p3nguin | vagonetas: Are you going to be asking it today? |
| 16:54.21 | leifmadsen | minotaur01: only way to do that is if your PC and Asterisk box are both plugged into the same hub (not switch) |
| 16:54.23 | VaGoNeTaS | what package or dependence do i need to install in order to have cdr_pgsql support? |
| 16:54.23 | ManxPower-work | You can capture the packets on your Asteriskbox using tcpdump, the copy the dump file to your PC and load it into wireshark. |
| 16:54.40 | VaGoNeTaS | i dont remember |
| 16:54.43 | VaGoNeTaS | is a lib |
| 16:54.43 | tzafrir_laptop | girlny, please run 'show modules like mysql' again. Are there any modules? |
| 16:54.44 | casix | tzafrir_laptop: I thing not intentionally. In /etc/asterisk/zaptel.conf they are defined: channel=>32-46,48-62 |
| 16:54.46 | VaGoNeTaS | its a lib |
| 16:54.47 | p3nguin | vagonetas: probably postgresql |
| 16:54.50 | VaGoNeTaS | no |
| 16:54.55 | *** join/#asterisk Tim_Toady (n=moi@77.49.24.242.dsl.dyn.forthnet.gr) |
| 16:54.57 | ManxPower-work | VaGoNeTaS: that depends on your distro. chances it would be something like postgresql-devel |
| 16:54.58 | VaGoNeTaS | pslib or something i dont remember |
| 16:55.10 | VaGoNeTaS | ubuntu 9.04 server |
| 16:55.14 | minotaur01 | ManxPower-work: i was hoping to be able to do it live |
| 16:55.15 | tzafrir_laptop | casix, is the '<' written there? |
| 16:55.24 | ManxPower-work | minotaur01: That will not be possible |
| 16:55.24 | leifmadsen | VaGoNeTaS: I think: sudo apt-get install pgsql-dev |
| 16:55.39 | tzafrir_laptop | VaGoNeTaS, apt-get build-depends asterisk |
| 16:55.40 | VaGoNeTaS | let me try |
| 16:55.54 | leifmadsen | VaGoNeTaS: otherwise: apt-cache search pgsql (or postgresql) |
| 16:56.00 | casix | tzafrir_laptop: yes it is .... :( |
| 16:56.04 | tzafrir_laptop | VaGoNeTaS, make sure you have deb-src sources for universe as well |
| 16:56.42 | tzafrir_laptop | casix, well, I figure you'll need to restart asterisk to fix that |
| 16:57.00 | tzafrir_laptop | notes that with newer versions 'dahdi restart' works fine |
| 16:57.33 | *** join/#asterisk joesuffceren (n=chatzill@ip68-104-167-226.ph.ph.cox.net) |
| 16:57.51 | joesuffceren | is "sip reload" from the cli sufficient for adding a new "localnet" entry, or do I need to restart asterisk |
| 16:57.59 | p3nguin | That's enough. |
| 16:58.06 | joesuffceren | thanks, p3nguin |
| 16:58.13 | casix | tzafrir_laptop: thanks |
| 16:58.51 | p3nguin | joesuffceren: Having NAT troubles? |
| 16:59.09 | VaGoNeTaS | libpq-dev |
| 16:59.11 | VaGoNeTaS | that is |
| 16:59.13 | VaGoNeTaS | thank you guys |
| 16:59.24 | VaGoNeTaS | c ya later |
| 16:59.25 | joesuffceren | p3nguin: no trouble. just adding a new vpn location |
| 16:59.31 | joesuffceren | so needed to add a new subnet |
| 17:00.43 | minotaur01 | for all the skeptics that said you cant do it: http://www.winpcap.org/docs/docs_40_2/html/group__remote.html#UNIX |
| 17:00.57 | girlny | tzafrir_laptop domU-12-31-39-01-7D-16*CLI> show modules like mysql |
| 17:00.57 | girlny | No such command 'show modules like mysql' (type 'help show modules like' for other possible commands) |
| 17:02.24 | p3nguin | girlny: module show like something |
| 17:04.53 | girlny | tzafrir_laptop domU-12-31-39-01-7D-16*CLI> module show like msql |
| 17:04.53 | girlny | Module Description Use Count |
| 17:04.53 | girlny | 0 modules loaded |
| 17:05.07 | girlny | my bad |
| 17:05.14 | tzafrir_laptop | mysql |
| 17:05.38 | p3nguin | girlny: and stop pasting shit in the channel. Use a pastebin if you're dying for us to see it. |
| 17:06.00 | dennis00 | Sorry, I had a phonecall, unfortunately not via VOIP. |
| 17:06.29 | dennis00 | But I am working with the BudgetPhone peer. |
| 17:06.31 | girlny | <PROTECTED> |
| 17:06.47 | ManxPower-work | sounds like girlny is using Asterisk 1.4 command syntax on Asterisk 1.6.x |
| 17:06.54 | dennis00 | Can somebody give me a hint in the right direction? |
| 17:07.09 | tzafrir_laptop | girlny, ok. So either those modules were not built or they were not installed |
| 17:07.25 | p3nguin | dennis00: Okay, the budgetphone peer... Let me look at you config in the pastebin for one moment. |
| 17:07.42 | tzafrir_laptop | in the asterisk-addons build directory: find * -name \*.so |
| 17:07.49 | tzafrir_laptop | does it show anything? |
| 17:08.01 | dennis00 | <PROTECTED> |
| 17:08.05 | p3nguin | dennis00: It sends calls to the [inbound] context. |
| 17:08.18 | p3nguin | dennis00: But I seriously doubt that the phone number they are sending to you is "1000" |
| 17:08.28 | ManxPower-work | It looks to me like your server is requesting auth password and your carrier is not providing it. |
| 17:09.09 | ManxPower-work | Based on the SIP debug the carrier is not sending any number, so the incoming call would match exten "s", but it really looks like the SIP debug is NOT COMPLETE. |
| 17:09.36 | dennis00 | extension = 1001, the phone number is 010 714.... |
| 17:09.41 | ManxPower-work | Normally SIP calls end with a BYE message. |
| 17:10.01 | p3nguin | dennis00: That's probably why it doesn't work. If they are sending your phone number, that is the exten that must match. |
| 17:10.26 | ManxPower-work | Too bad it took so long to get the info. [TK]D-Fender must have gotten sick of waiting and went back to doing productive stuff. |
| 17:10.42 | p3nguin | dennis00: Does that make sense? |
| 17:10.54 | dennis00 | p3nguin: the extension must be my phone number? |
| 17:11.01 | ManxPower-work | p3nguin: according the the SIP debug he pasted the number is "s" |
| 17:11.12 | dennis00 | are you positive? |
| 17:11.13 | ManxPower-work | p3nguin: did you look at his sip debug pastebin? |
| 17:11.16 | p3nguin | dennis00: If they are SENDING YOUR PHONE NUMBER, the exten that you must match is your phone number. |
| 17:11.20 | p3nguin | manxpower-work: no |
| 17:11.27 | dennis00 | the number is 1001 in my extensions.conf |
| 17:11.30 | ManxPower-work | p3nguin: go look, notice the s@ipaddress |
| 17:11.43 | p3nguin | I don't give a rat's ass what the number in your extensions.conf is. |
| 17:11.47 | ManxPower-work | dennis00: we don't really care about extension 1000. |
| 17:11.52 | p3nguin | exactly |
| 17:12.00 | p3nguin | If you're doing it wrong, it's a worthless thing. |
| 17:12.01 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 17:12.27 | ManxPower-work | But what does seem important is "INVITE sip:s@93.186.178.15 SIP/2.0" |
| 17:13.15 | dennis00 | exten => 1001,1,Answer() < so what should this 1001 be? |
| 17:13.22 | p3nguin | deleted |
| 17:13.42 | p3nguin | If nothing sends the phone number "1001" into that context, delete it. |
| 17:13.50 | ManxPower-work | dennis00: it does not matter at all about the 1000 or 1001 because you have a much lower level problem that must be solved before those extensions do anything but distract you from our questions |
| 17:14.02 | tzafrir_laptop | girlny, any such mysql module there? |
| 17:14.12 | dennis00 | ManxPower-work: I see. |
| 17:14.26 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 17:14.26 | *** mode/#asterisk [+o file] by ChanServ |
| 17:14.26 | dennis00 | ManxPower-work: But to be honest, I have no idea how to troubleshoot here. |
| 17:14.29 | ManxPower-work | You are asking about a flat tire when your car doesn't even have an engine. Then you are continuing to ask about your flat time when we tell you it does not matter until you have an engine in your call. |
| 17:15.15 | ManxPower-work | dennis00: try another call with sip debug enabled and wait for the CLI to stop printing debug messages THEN pastebin the into |
| 17:15.22 | dennis00 | I have a sip for budgetphone, that points to [inbound] which points to 1001... which is configured in my SIP-device, X-lite and iPhone. |
| 17:15.26 | *** join/#asterisk Qwell (i=north@pdpc/sponsor/digium/Qwell) |
| 17:15.26 | *** mode/#asterisk [+o Qwell] by ChanServ |
| 17:15.32 | ManxPower-work | dennis00: NO!@ |
| 17:15.36 | p3nguin | dennis00: Sorry, but that is not how it works. |
| 17:15.44 | ManxPower-work | "that points to [inbound] which points to 1001" <- this is not true. |
| 17:15.47 | p3nguin | dennis00: Forget whatever you think you know about contexts. |
| 17:15.53 | girlny | tzafrir_laptop no modules with mysql |
| 17:16.01 | p3nguin | dennis00: extens are MATCHES |
| 17:16.15 | p3nguin | dennis00: If the number does not MATCH your exten, then it does not apply. |
| 17:16.37 | dennis00 | and this is my phone number or the phone number I am ringing? |
| 17:17.26 | *** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb) |
| 17:17.26 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:17.31 | p3nguin | You need to match whatever the ITSP is sending to you when you call your DID. |
| 17:17.34 | ManxPower-work | dennis00: You have sip from budgetphone, the budgetphone peer in sip.conf points to the inbound context. The call comes in, matches the sip peer, gets send to whatever destination number your carrier sent the call to. That destination number much have a match in the inbound context or the call will be rejected |
| 17:17.59 | ManxPower-work | But you see, the call isn't even getting into extensions.conf based on the SIP debug you posted. |
| 17:18.16 | dennis00 | I see, that's odd. |
| 17:18.20 | ManxPower-work | In any case, I give up. |
| 17:18.27 | ManxPower-work | I wish you the BEST of luck, dennis00 |
| 17:18.46 | *** join/#asterisk oldhack (n=jfincher@cpe-24-27-56-221.austin.res.rr.com) |
| 17:19.13 | dennis00 | :( ok. |
| 17:20.49 | dennis00 | http://pastebin.ca/1751190 here is the full debug log. |
| 17:21.52 | p3nguin | Is that from an off-site phone TO YOUR BUDGETPHONE DID? |
| 17:22.29 | dennis00 | Yes. |
| 17:23.14 | p3nguin | # |
| 17:23.14 | p3nguin | From: "asterisk" <sip:asterisk@93.186.178.15>;tag=as54a02b91 |
| 17:23.14 | p3nguin | # |
| 17:23.15 | p3nguin | To: <sip:1001@127.0.0.1> |
| 17:23.23 | p3nguin | fucking pastebin.ca |
| 17:23.44 | p3nguin | I hate how it adds hash marks to my copying. |
| 17:23.57 | p3nguin | dennis00: In that, which one are you? |
| 17:24.07 | tzafrir_laptop | girlny, rpm -qa | grep mysql |
| 17:24.11 | dlynes | p3nguin, i hate how it adds line numbers to stuff I copy out of it into my editor :( |
| 17:24.22 | dennis00 | 1001 is my iPhone, Voipbuster is my X-Lite |
| 17:24.26 | p3nguin | dlynes: Use pastebin.com instead. |
| 17:24.34 | p3nguin | dennis00: I don't care. |
| 17:24.38 | dlynes | p3nguin, as opposed to what? |
| 17:24.49 | p3nguin | dlynes: pastebin.ca like we were talking about. |
| 17:24.55 | girlny | [root@domU-12-31-39-01-7D-16 asterisk-addons-1.6.0.4]# rpm -qa | grep mysql |
| 17:24.55 | girlny | mysql-libs-5.0.45-6.fc8 |
| 17:24.55 | girlny | mysql-server-5.0.45-6.fc8 |
| 17:24.55 | girlny | mysql-5.0.45-6.fc8 |
| 17:24.55 | girlny | php-mysql-5.2.6-2.fc8 |
| 17:24.55 | dlynes | p3nguin, oh |
| 17:24.59 | dennis00 | p3nguin: which one am I? what do you mean? |
| 17:25.10 | p3nguin | girlny: WHAT IS YOUR PROBLEM? |
| 17:25.20 | p3nguin | girlny: WHY CAN'T YOU USE A GODDAMNED PASTEBIN? |
| 17:25.26 | p3nguin | girlny: ARE YOU A RETARD? |
| 17:25.56 | p3nguin | <PROTECTED> |
| 17:25.59 | p3nguin | To: <sip:1001@127.0.0.1> |
| 17:26.11 | p3nguin | dennis00: Which one are you? |
| 17:26.11 | dennis00 | 93.186 = my server |
| 17:26.17 | dennis00 | 127.0.0.1 = my iphone |
| 17:26.31 | *** join/#asterisk jamicque (i=jamicque@jam.bema.one.pl) |
| 17:26.35 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 17:26.57 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 17:26.57 | *** mode/#asterisk [+o file] by ChanServ |
| 17:27.35 | jamicque | hi @ll, I have a problem with mwi on asterisk (1.6.1.12) LINKSYS SPA phone subscirbes for the mailbox, however no futher messages from asterisk to phone are sned about new voicemail. Can anyone help me? |
| 17:28.16 | p3nguin | dennis00: What is your budgetphone DID that I can dial from my cell phone? |
| 17:28.39 | dlynes | p3nguin, only pastebin.ca does that? I had stopped using pastebin.com a long time ago because it always timed out on me, and pastebin.ca syntax highlighted code |
| 17:29.05 | p3nguin | In years past, pastebin.com was slow as could be. |
| 17:29.17 | p3nguin | After people started using other paste services, it got better. |
| 17:29.21 | dennis00 | +31107142866 |
| 17:29.38 | dlynes | ah...I guess it was just too bogged down...didn't have enough bandwidth |
| 17:29.53 | p3nguin | They do have different highlighting than .ca does, though. |
| 17:30.13 | dlynes | p3nguin, oh...last time i used it, .com didn't have highlighting at all |
| 17:30.23 | *** join/#asterisk HenrikJott (n=info@d83-183-134-141.cust.tele2.se) |
| 17:30.27 | p3nguin | That must have been 10 years ago! |
| 17:30.33 | dlynes | nah |
| 17:30.40 | dlynes | only maybe 3 to 5 years ago |
| 17:31.07 | dlynes | Yeah...probably more like 4 to 5 years ago |
| 17:31.22 | dlynes | Around the time I first started using asterisk (late 1.0 series to early 1.2 series) |
| 17:32.36 | dlynes | jamicque, do you have the voicemail server ip/hostname set to the same as the proxy server ip/hostname? |
| 17:32.36 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.218.101.dsl.dyn.forthnet.gr) |
| 17:32.45 | girlny | . |
| 17:32.57 | HenrikJott | Hi all! I have a problem with the ring-strategy in asterisk. i´m currently using the leastrecent-strategy in my queue and that way the agent that got a call least recently from the queue gets the highest priority. I´m wondering if there is a way to make asterisk set the highest priority to the agent who least recently hung up a call? |
| 17:33.01 | dennis00 | p3nguin: +31107142866, is that what you mean by did? |
| 17:33.01 | dlynes | jamicque, also, do you have the 'mailbox=' line in your sip.conf file set to point to the correct entry in your voicemail.conf file? |
| 17:33.20 | p3nguin | dennis00: Change your register statement to 31107142866:budgetphonepassword@sip1.budgetphone.nl/31107142866 |
| 17:33.25 | p3nguin | dennis00: yes, that was what I meant. |
| 17:33.45 | dennis00 | p3nguin: their forum says no /31107142866. But posts in 2005 say with /31107142866. But I will try. |
| 17:33.50 | p3nguin | dennis00: Change it to exactly what I typed, save the file, reload sip. |
| 17:34.54 | p3nguin | (when I say exactly, I mean use your real password if it is different) |
| 17:35.07 | dennis00 | I am not that stupid ^^, but I have done it. |
| 17:35.25 | p3nguin | I just wanted to be clear when I said "exactly." |
| 17:35.28 | jamicque | dlynes I use realtime. I think that the configuration is correct |
| 17:35.29 | dennis00 | I can still send outgoing calls, but still no incoming calls. |
| 17:35.39 | p3nguin | dennis00: Did you do what I wanted? |
| 17:35.51 | dennis00 | p3nguin: Yes, I changed sip.conf and restarted asterisk. |
| 17:35.59 | p3nguin | dennis00: Now show me a debug of a new call from an off-site phone calling your DID. |
| 17:36.26 | p3nguin | dennis00: Not part of one, but an entire debug from start to finish. |
| 17:36.43 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 17:37.03 | dennis00 | http://pastebin.ca/1751216 |
| 17:37.37 | p3nguin | That's not the entire call. |
| 17:38.30 | dennis00 | http://pastebin.ca/1751219 |
| 17:38.33 | p3nguin | That might be what happend when you called, but that is not a call. If that makes sense. |
| 17:38.40 | dennis00 | Yes. |
| 17:39.48 | *** join/#asterisk sgimeno (n=santiago@226.Red-80-33-64.staticIP.rima-tde.net) |
| 17:40.12 | p3nguin | The thing standing out to me is No user 'Unknown' in SIP users list Found peer '31107142866' for 'Unknown' from 83.143.188.165:5060 |
| 17:40.40 | dennis00 | Hmm,, maybe it goes to the default? Or maybe testing via voipbuster is not a good idea. |
| 17:41.01 | *** join/#asterisk nny (n=scott@64.203.239.83) |
| 17:41.07 | p3nguin | How can you test budgetphone via voipbuster? |
| 17:41.15 | nny | any Cisco 50Xg users here? Need someone to check something |
| 17:41.37 | nny | something = sidetone, this phone seems to lack it right now, where my SPA962 doesn't. |
| 17:41.40 | dennis00 | I call the budgetphone number while connected to the voipbuster SIP. |
| 17:41.50 | p3nguin | Voipbuster is an on-site configuration, which I specifically said to use an off-site phone. |
| 17:42.10 | dennis00 | and the iPhone is connected to the SIP of my own Asterisk. If I connect it to sip1.budgetphone.nl via the iPhone, I can be called via 010 number. |
| 17:42.11 | p3nguin | Such as pick up your cell phone and dial 31107142866. |
| 17:42.36 | dennis00 | I don't have a cell phone on the cellular network until next Monday. |
| 17:43.03 | girlny | tzafrir_laptop any clues what should i try next |
| 17:43.19 | *** join/#asterisk acclaimtech (n=kevin@70.62.14.66) |
| 17:43.23 | acclaimtech | Finally |
| 17:43.38 | dennis00 | I have been able to call 1001 or my phone number via my own SIP, but not via Voipbuster or any external phone. |
| 17:43.39 | acclaimtech | That whole verification process with nickserv takes too long. |
| 17:43.52 | ManxPower-work | dennis00: and that will mix the incoming and outgoing sip traffic in your debug. Not very productive, useful, or helpful. |
| 17:44.12 | dennis00 | ManxPower-work: I am only connected to my own SIP via 1 node. |
| 17:44.35 | acclaimtech | Can Asterisk be used from a ESXi Vsphere virtual machine? |
| 17:44.38 | dennis00 | But I think I understand what you mean, if I dont use voipbuster to test. |
| 17:45.04 | dennis00 | acclaimtech: Asterisk can be used on a VPS, but I hear a lot of people hate to use a VPS for that. |
| 17:45.46 | acclaimtech | dennis what exactly do I need to get rid of my vonage and just build my own asterisk service for 2 lines? |
| 17:46.44 | dennis00 | p3nguin: the order of includes does not pose a problem? |
| 17:46.54 | p3nguin | At this point, no. |
| 17:48.11 | HenrikJott | Does anyone know how to make the "leastrecent" ring strategy check which agent was least recently diconnected from a call instead of least recently called? |
| 17:50.54 | dennis00 | p3nguin: When I dial the number, sometimes it keeps on saying 'dialing' for 5 seconds, before it gives the message in Dutch. |
| 17:51.15 | p3nguin | dennis00: What does "it" mean? |
| 17:51.21 | EyeBaw | Whats wrong with being Dutch? |
| 17:51.34 | dennis00 | it => woman. |
| 17:52.10 | p3nguin | No, "it" is something else. I'm so fed up with people being cryptic or not describing things where they can be understood. |
| 17:55.17 | *** join/#asterisk [netman] (n=netman@17.Red-81-36-134.dynamicIP.rima-tde.net) |
| 17:55.53 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 17:56.19 | acclaimtech | Is there a guide on what hardware you specifically need to setup asterisk? |
| 17:56.54 | acclaimtech | all I need is two lines and I'm not looking to make a significant hardware investment |
| 17:56.58 | p3nguin | It pretty much runs on anything. |
| 17:57.08 | p3nguin | What do you mean by "lines?" |
| 17:57.20 | acclaimtech | phone lines |
| 17:57.25 | acclaimtech | I need a voice line and a fax line |
| 17:57.27 | p3nguin | Be more specific. |
| 17:57.53 | acclaimtech | two different phone numbers |
| 17:58.00 | p3nguin | Still not helpful. |
| 17:58.17 | p3nguin | Maybe an example will help you... |
| 17:58.20 | acclaimtech | I want to replace my vonage unit |
| 17:58.25 | acclaimtech | it has two lines |
| 17:58.32 | p3nguin | "I need a SIP service that will provide me with two phone numbers." |
| 17:58.43 | acclaimtech | there you go |
| 17:59.00 | acclaimtech | is there a way to get a SIP permenantly? |
| 17:59.03 | *** join/#asterisk agol (n=kvirc@87.69.47.202.cable.012.net.il) |
| 17:59.07 | acclaimtech | without having any monthly payment? |
| 17:59.08 | p3nguin | Then all you need is a computer good enough to run Linux and Asterisk,. |
| 17:59.21 | acclaimtech | ok |
| 17:59.35 | acclaimtech | so then what do use a modem to connect the phone lines to? |
| 17:59.39 | acclaimtech | or do you use a pbx |
| 17:59.43 | p3nguin | There are at least three ITSPs that I know of that will give you a free DID (phone number that people dial to reach you). |
| 18:00.18 | p3nguin | sipgate, ipkall, and ipcomms |
| 18:00.22 | carrar | great money making model |
| 18:00.48 | p3nguin | They will give you free phone numbers for incoming calls. If you want to make outgoing calls, you'll need someone to give you termination service. |
| 18:01.06 | p3nguin | To connect regular phone to your asterisk system, you'll need a ATAs. |
| 18:01.08 | p3nguin | ~ata |
| 18:01.09 | infobot | methinks ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA |
| 18:01.28 | p3nguin | Alternatively, you can use IP phones and connect them all by Ethernet. |
| 18:01.57 | p3nguin | Alternative to that, you can use free software phones on computers which are network-enabled. |
| 18:02.04 | casix | bye |
| 18:02.11 | acclaimtech | ah |
| 18:02.14 | acclaimtech | what about faxing? |
| 18:02.33 | p3nguin | Faxing over SIP is something most people don't want to deal with. |
| 18:02.48 | p3nguin | It is possible to do it, but it is not reliable. |
| 18:02.53 | titter | I have had success with fax via my PRI's |
| 18:03.10 | p3nguin | FAX is already a compressed transmission, so cramming it over VoIP causes additional problems. |
| 18:03.21 | acclaimtech | ah |
| 18:03.29 | p3nguin | Fax over PRI has much more reliability. |
| 18:03.29 | acclaimtech | so I should just stick with Vonage then? |
| 18:03.49 | p3nguin | Does Vonage allow faxing? |
| 18:03.54 | acclaimtech | yep |
| 18:04.05 | p3nguin | That's interesting to know. I had no idea. |
| 18:04.11 | acclaimtech | I have two lines a voice line and a fax line using vonage |
| 18:04.14 | titter | I have also had success with fax over my SIP provider ... but like said, it is not that reliable ... it works, and works well, but often times the call will drop |
| 18:04.54 | acclaimtech | I was just trying to find a way to get rid of the 25 dollar a month bill. |
| 18:04.57 | p3nguin | I have been able to get a couple faxes over SIP, but I think I have a problem with my internal configuration. |
| 18:05.17 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 18:05.29 | p3nguin | I haven't spent $25 on phone service in three months. |
| 18:05.46 | titter | I think my issue is from my SIP provider ... they hangup the call, didn't really look into it much as my main concern was making sure it worked on my PRI. I just used my home server for testing since it doesnt matter if it went down. |
| 18:05.50 | p3nguin | More like $15 in that three months. |
| 18:06.35 | p3nguin | You could use sipgate and let them take care of the faxes within them reaching your box. |
| 18:06.48 | p3nguin | I don't know how much they charge for the fax add-on, though. |
| 18:07.02 | muiro | is there any way I can make Read or WaitExten wait for an indefinite amount of time? |
| 18:07.19 | titter | ya the main thing for my company was to eliminate all e-fax accounts ... we were getting charged per page, and some of the bills were in the thousands per month |
| 18:08.06 | p3nguin | muiro: Set the wait values to something extremely high. |
| 18:08.17 | muiro | p3nguin: alright |
| 18:08.26 | p3nguin | I think WaitExten() will wait for hours. |
| 18:08.35 | p3nguin | without giving any value at all. |
| 18:09.00 | titter | i used this http://caspian.dotconf.net/menu/Software/SendEmail/ to send the fax after converting with tiff2pdf. i tried other ways, the fax2mail script on voip-info (didn't like the way it worked), then just a System() command via mailx ... but it would complain about the | in my command of echo "message" | mail -a file -s subject emial@email.com ... so I settled with that perl script |
| 18:09.27 | p3nguin | I probably wouldn't want someone to wait for timeout forever, though. I would set the timers to something like 2 minutes at most. |
| 18:09.29 | muiro | my dialplan requires autofallthrough, so I'll just have to pump it with an unreasonable number of seconds |
| 18:09.31 | *** join/#asterisk jmayorga5 (n=jmayorga@adsl-99-91-103-186.dsl.ipltin.sbcglobal.net) |
| 18:09.49 | p3nguin | 65535? |
| 18:10.10 | p3nguin | You might be able to add a couple together, as well. |
| 18:10.22 | jmayorga5 | Can anybody point me to a good resource on implementing the jitter buffer in Asterisk 1.6.x? |
| 18:10.46 | p3nguin | WaitExten() followed by another WaitExten() would double whatever timeout it had by default. |
| 18:10.49 | muiro | p3nguin: well, I needed to add some extra functionality to MeetMe, and rather than writing a module or something, just doing it in dialplan by originating a WaitExten into the conference and passing dtmf through |
| 18:11.11 | acclaimtech | hmmm |
| 18:11.16 | muiro | p3nguin: I need it to be running for the length of a given conference, anywhere from like 10 minutes to an hour |
| 18:11.34 | ManxPower-work | muiro: no dialplan requires autofallthru |
| 18:12.00 | verywiseman | i have tdm2400P, with 6 fxo module, am i write chan_dahdi.conf for than module as i write them in zapata.conf ? |
| 18:12.27 | [TK]D-Fender | veyMostly. |
| 18:12.29 | muiro | ManxPower-work: the way I have other things written work better when I have autofallthrough. Though, I have recoded all of it since I put that in... |
| 18:12.44 | [TK]D-Fender | verywiseman: Mostly. Go read the migration docs in the tarball |
| 18:12.59 | acclaimtech | so is there a way to provide your own service and not have a service provider? |
| 18:13.33 | muiro | I'm getting to the point where I'm going to have to write my own modules, I'm just being extremely stubborn and putting stuff in the dialplan that's way too complex to be there |
| 18:14.20 | muiro | there will eventually be a harrowing, lol |
| 18:14.39 | p3nguin | acclaimtech: The service providers typically just give you origination (calls from the PSTN to you) and termination (calls from you to the PSTN). If you don't need to make those types of calls, you don't neen an ITSP. |
| 18:16.32 | acclaimtech | p3nguin: I want regular phones and so on to be able to call me and me to call them |
| 18:17.25 | bmoraca_work | acclaimtech, then you need a service provider |
| 18:17.50 | *** join/#asterisk coppice (n=chatzill@106.202.17.210.dyn.pacific.net.hk) |
| 18:18.28 | acclaimtech | what does the service provider have for equipment that allows them to make the conversion? |
| 18:19.30 | p3nguin | VoIP-to-PSTN and PSTN-to-VoIP gateways |
| 18:20.06 | p3nguin | You can have those, too, but you'll have to retain a connection to the telco. |
| 18:20.13 | acclaimtech | ok p3nguin so how much is one of those? |
| 18:20.15 | acclaimtech | ah |
| 18:20.20 | acclaimtech | just one connection? |
| 18:20.27 | p3nguin | Depends on the type of connection. |
| 18:20.44 | acclaimtech | Well let's say I wanted to become a sip provider for my customers |
| 18:20.46 | p3nguin | With a T1 PRI, I think you can handle 23 phone calls. |
| 18:21.16 | p3nguin | SOmeone with T1 and PRI knowledge would have to say if that is the right number or not. |
| 18:21.27 | p3nguin | I do all IP. No PSTN connection at all. |
| 18:21.32 | voipmonk | yes |
| 18:21.34 | voipmonk | 23 calls |
| 18:21.37 | voipmonk | 23 b channels |
| 18:21.38 | voipmonk | 1 d channel |
| 18:21.54 | acclaimtech | and you need a t1 to do this? |
| 18:22.03 | acclaimtech | so you have to have a circuit |
| 18:22.03 | voipmonk | u dont need a t1 for 23 calls |
| 18:22.08 | p3nguin | For that specific type of configuration, yes. |
| 18:22.17 | acclaimtech | could you use cable modems? |
| 18:22.38 | p3nguin | You can do some calls with cable modems, but you won't have PSTN connectivity. |
| 18:22.54 | voipmonk | wow... |
| 18:23.42 | acclaimtech | hmmmmm..... |
| 18:23.49 | acclaimtech | that won't help |
| 18:24.41 | *** part/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 18:24.58 | voipmonk | so very misinformed people are, yes... wow |
| 18:25.15 | acclaimtech | so then what is the advantage of using asterisk? |
| 18:25.17 | titter | unless you have a voip service with your cable provider and use an analog card to connect to their modem ... but that's pretty limited for multi lines |
| 18:25.46 | *** join/#asterisk MACscr (n=Mark@c-98-214-100-212.hsd1.il.comcast.net) |
| 18:26.12 | MACscr | when using a softphone, the big negative aobut it is that you can't really doing any QOS on it, right? |
| 18:26.25 | voipmonk | no |
| 18:27.30 | MACscr | grr, I love my softphone, but I can't be having poor quality,etc, at times because my pc at the time is working hard at something, etc |
| 18:27.34 | titter | acclaimtech: it's robust, and free. if you want to be a sip provider you will need some sort of PSTN connectivity ... and bandwidth to handle the IP traffic |
| 18:27.36 | acclaimtech | So either way in the US it all goes back to AT&T |
| 18:27.37 | voipmonk | your soundcard and other apps interfering with the audio that must reach the other persons ear in the right sequence at the right time in order for it to sound like something the other person can comprehend... get an ip phone, take a day or so to understand wtf is going on in your network before you deploy voip |
| 18:28.07 | voipmonk | read a little |
| 18:28.11 | voipmonk | learn a lot |
| 18:28.40 | titter | acclaimtech: not really ... |
| 18:29.10 | MACscr | voipmonk: lol, I already a polycom 330. I just prefer the actual day to day use of a softphone. One headset for everything. Easy copy and paste of phone numbers, etc |
| 18:29.26 | *** join/#asterisk lynxsys (n=thelynx@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 18:30.11 | acclaimtech | But asterisk is a Virtual pbx right? |
| 18:30.21 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 18:30.24 | acclaimtech | so for a company you can have auto attendant and so on? |
| 18:31.27 | p3nguin | Asterisk is an actual PBX... what would be virtual about it? |
| 18:31.41 | titter | ^ what he said |
| 18:31.43 | voipmonk | yes acclaimtech |
| 18:31.58 | voipmonk | what do u want to do with asterisk that has brought you here, acclaimtech |
| 18:32.05 | p3nguin | Yes, you can create attendants and IVRs with asterisk. |
| 18:32.26 | acclaimtech | I wanted to get rid of my vonage bill and just have a one time hardware cost for two phone lines including a fax line |
| 18:32.44 | acclaimtech | and run it off the ESX server |
| 18:32.58 | titter | you will need some sort of connection to a telco |
| 18:33.28 | acclaimtech | right... |
| 18:33.43 | acclaimtech | I understand now. This is more for internal office extensions and so forth |
| 18:34.02 | titter | not really, I do what you want at home |
| 18:34.30 | titter | I use a sip provider, and connect to them with a sip trunk ... they give me my two phone numbers |
| 18:34.31 | acclaimtech | so how do you connect to the telco? |
| 18:34.37 | acclaimtech | ah |
| 18:34.41 | acclaimtech | how much is that monthly? |
| 18:35.18 | titter | depends on the provider ... one number is incoming only and it's 5/mo, the other is unlimited national and its 15/mo |
| 18:35.26 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 18:35.32 | titter | in/out on the 2nd number |
| 18:35.32 | voipmonk | i pay around $50/mo for 4 families |
| 18:35.37 | acclaimtech | so it's about the same as vonage? |
| 18:35.44 | titter | my provider yes |
| 18:35.47 | titter | there are cheaper ones |
| 18:35.48 | voipmonk | 3 inbound numbers and a crap load of talking for 4 families |
| 18:36.04 | carrar | in one house? |
| 18:36.16 | voipmonk | no |
| 18:36.16 | carrar | heh |
| 18:36.16 | voipmonk | 4 different homes |
| 18:36.23 | *** join/#asterisk rocksfrow (n=kyle@74-92-146-97-WashingtonDC.hfc.comcastbusiness.net) |
| 18:36.26 | voipmonk | im the phone guy for my fam |
| 18:36.27 | acclaimtech | well I need to justify the monthly cost |
| 18:36.40 | titter | acclaimtech: does your cable provider have voip services? |
| 18:36.50 | acclaimtech | they have digital phone |
| 18:36.57 | voipmonk | thats voipp |
| 18:36.59 | EyeBaw | My cable provider gives me networking cable for fre |
| 18:37.00 | voipmonk | voip |
| 18:37.05 | EyeBaw | I use this to plug into the mainframe |
| 18:37.17 | voipmonk | digital phone is another marketing brainwashing name given to VoIP |
| 18:37.19 | fenrus | a mainframe? :) |
| 18:37.21 | acclaimtech | right |
| 18:37.29 | acclaimtech | I know that voipmonk |
| 18:37.29 | titter | i use my comcast "digital voice" with an analog card since their modem becomes my FXS |
| 18:37.31 | fenrus | ibm s390? (: |
| 18:37.36 | MACscr | acclaimtech: if I were you, I wouldn't consider doing fax over sip |
| 18:37.42 | EyeBaw | Yeah :) |
| 18:38.06 | MACscr | keep a pstn line for that |
| 18:38.12 | p3nguin | acclaimtech: I pay $1/mo for my toll-free phone number, and 2.9 cents/minute for the incoming calls on it. It is delivered to me over IAX2. |
| 18:38.51 | acclaimtech | well... After talking to you in this room I had another idea of just getting a bonded t1 and install an asterisk box and then hand out numbers to our clients |
| 18:38.56 | acclaimtech | and become a sip provider |
| 18:39.06 | p3nguin | acclaimtech: Look at flowroute.com and voip.ms for rates. |
| 18:39.32 | p3nguin | VoIP.ms even provides you with a reseller's panel. |
| 18:39.37 | MACscr | acclaimtech: why are you trying to become a sip provider if you don't have much experience with asterisk yet, etc? |
| 18:39.40 | acclaimtech | a bonded t should be able to hand about 40 calls right? |
| 18:39.51 | titter | acclaimtech: don't forget about long distance costs |
| 18:40.20 | Kobaz | acclaimtech: depends how many are bonded |
| 18:40.30 | MACscr | gotta get did's of course as well |
| 18:40.45 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 18:40.58 | Kobaz | p3nguin: voip.ms uses vitelity for some of their network |
| 18:41.08 | Kobaz | although the reseller panel is pretty nifty |
| 18:41.20 | p3nguin | I know. |
| 18:42.05 | MACscr | acclaimtech: a decent bonded t1 is going to cost you about $600 - $1000, plus your hardware costs, plus did's, plus long distance. Your profit is going to be slim, if not nil |
| 18:42.16 | MACscr | plus of course your time |
| 18:42.51 | titter | on average my pri circuits cost me around 650/mo |
| 18:43.05 | p3nguin | I think you'll be better off buying a case of ATAs and being a VoIP.ms reseller to your clients. |
| 18:44.05 | ManxPower-work | acclaimtech: that all depends on your codec |
| 18:44.46 | MACscr | remember its going to be two connections for each call as well, incoming/outgoing |
| 18:47.55 | rocksfrow | is the only reason in buying a card with multiple spans to support more numbers? |
| 18:48.33 | ManxPower-work | rocksfrow: "more numbers"? |
| 18:48.49 | rocksfrow | phone numbers sorry |
| 18:48.53 | rocksfrow | DIDs? |
| 18:49.04 | ManxPower-work | rocksfrow: You can put several thousand numbers on one PRI. |
| 18:49.07 | rocksfrow | basically i'm just wondering if it's worth investing in a card with an extra span |
| 18:49.11 | rocksfrow | oh you can?? |
| 18:49.20 | rocksfrow | so what is the purpose in multiple spans? |
| 18:49.20 | ManxPower-work | You could only have up to 23 calls ACTIVE, but you can have thousands of numbers. |
| 18:49.26 | rocksfrow | ohhh right okay |
| 18:49.29 | rocksfrow | channels is what i meant i believe? |
| 18:49.49 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.250.dyn.user.ono.com) |
| 18:49.52 | ManxPower-work | A single channel can handle a single call. You would want more spans if you needed to have more calls active at the same time. |
| 18:49.58 | rocksfrow | okay so does that answer my question then? multiple spans are for supporting more active calls? |
| 18:50.06 | ManxPower-work | yes. |
| 18:50.18 | rocksfrow | okay sweet thanks man |
| 18:50.34 | ManxPower-work | rocksfrow: on a PRI, a call comes in on any channel and the telco send the dialed number so the PBX knows what DID was dialed. |
| 18:50.48 | rocksfrow | right. |
| 18:50.51 | rocksfrow | the channel is random, right |
| 18:51.02 | ManxPower-work | rocksfrow: on other technologies like CAS T-1 or analog you generally don't know what number was dialed so you can only have one number per channel. |
| 18:51.11 | rocksfrow | yeaa see thats my issue |
| 18:51.14 | rocksfrow | i'm actually switching to PRI |
| 18:51.23 | rocksfrow | i'm currently on analog, and limited with no DID |
| 18:51.31 | ManxPower-work | rocksfrow: PRIs have sub-seconds call setup/teardonw |
| 18:51.50 | carrar | much faster |
| 18:51.52 | rocksfrow | we're actually saving ~$200/mo on our bill now too! |
| 18:51.55 | ManxPower-work | so by the time your finger lifts off the last digit the all is already being processed |
| 18:52.41 | *** join/#asterisk MACscr (n=Mark@c-98-214-100-212.hsd1.il.comcast.net) |
| 18:52.43 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 18:54.00 | rocksfrow | who's your favorite digium outlet? :-p |
| 18:54.12 | rocksfrow | \/distributor |
| 18:54.14 | ManxPower-work | rocksfrow: Sangma |
| 18:54.22 | voipmonk | Sangoma |
| 18:54.23 | ManxPower-work | I don't use Digium cards anymore. |
| 18:54.40 | voipmonk | i havent made that leap yet :) |
| 18:54.42 | *** join/#asterisk corretico (n=laguilar@201.201.46.106) |
| 18:54.46 | rocksfrow | interesting |
| 18:54.50 | rocksfrow | how come? |
| 18:54.53 | girlny | can anyboby help me with this please domU-12-31-39-01-7D-16*CLI> cdr mysql status ---- Not currently connected to a MySQL server. |
| 18:55.08 | voipmonk | Xen |
| 18:55.11 | carrar | use postgres! |
| 18:55.35 | ManxPower-work | rocksfrow: many, many, many people have NO trouble with Digium cards. I am not one of those people. |
| 18:56.02 | ManxPower-work | carrar: girlny is a total linux n00b |
| 18:56.17 | carrar | heh |
| 18:57.20 | girlny | lol you right |
| 18:57.43 | girlny | any clues on how solve that error |
| 18:58.02 | verywiseman | how can i know if fxo channels is defined in asterisk? |
| 18:58.41 | carrar | is your db running? |
| 18:58.41 | carrar | is your db setup correct? |
| 18:58.41 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 18:59.00 | ManxPower-work | verywiseman: "zap show channels" or "dahdi show channels" |
| 18:59.06 | *** join/#asterisk mrchrisadams (n=mrchrisa@87-194-125-43.bethere.co.uk) |
| 19:00.24 | girlny | yes the db is fine |
| 19:00.51 | carrar | check your asterisk config and make sure the module is loaded |
| 19:01.32 | girlny | what perticular module has to be loaded |
| 19:02.05 | carrar | cdr_addon_mysql |
| 19:02.19 | carrar | something like that |
| 19:02.54 | girlny | well i added that to the module.conf |
| 19:03.10 | carrar | is it loaded? |
| 19:03.34 | girlny | sorry... but how do i check |
| 19:03.41 | carrar | module show |
| 19:08.25 | girlny | yes it say 3 modules loaded when i put "module show like mysql" cdr_addon_mysql.so res_config_mysql.so app_addon_sql_mysql.so |
| 19:08.57 | carrar | then take a look at your mysl config file in /etc/asterisk |
| 19:09.07 | carrar | make sure it's correct |
| 19:09.25 | telnettech | girlny: can you log into your db from the Linux prompt manually? |
| 19:10.25 | girlny | this is all i get autoclear=yes |
| 19:10.39 | carrar | All you ger from what? |
| 19:10.50 | carrar | connecting from the cli? |
| 19:11.46 | telnettech | from where girlny |
| 19:12.42 | carrar | heh |
| 19:13.09 | girlny | in mysql.conf it only has autoclear=yes |
| 19:13.14 | girlny | inside of it |
| 19:13.26 | HenrikJott | I have one more question! I have read that G.711 codec consumes about 87kbps, now i wonder if that is one-direction or two-way? |
| 19:13.35 | carrar | read up on how to configure mysql.conf |
| 19:14.42 | *** join/#asterisk SMcCormack (n=samccorm@astound-66-234-210-133.ca.astound.net) |
| 19:14.44 | [TK]D-Fender | Heretic: Per direction |
| 19:14.46 | telnettech | girlny: you need to setup the mysql.conf file so that Asterisk knows where, and how to connect to the DB....it doesnt go out and look for it cause you have the module running |
| 19:15.00 | [TK]D-Fender | Heretic: 87 in + 87 out |
| 19:15.14 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 19:15.26 | HenrikJott | [TK]D-fender: ahh, okay, then my traffic calculations seems a lot more correct! =) thanks! |
| 19:17.10 | SMcCormack | Anyone on here managed to get Asterisk to compile on a mac running 10.6? |
| 19:17.48 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:19.02 | girlny | carrar do you mean cdr_mysql.conf |
| 19:21.11 | carrar | yes |
| 19:21.23 | carrar | this is the cdr loggin you are trying to fix correct? |
| 19:22.35 | carrar | You connecting to a db on the server or on some other server? |
| 19:22.41 | *** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131) |
| 19:22.51 | *** join/#asterisk Laureano (n=Laureano@190.245.116.231) |
| 19:23.07 | Laureano | Is it viable to use Zaptel 1.4 with Libpri 1.2 and Asterisk 1.2? |
| 19:23.39 | [TK]D-Fender | Laureano: Not possible |
| 19:23.50 | carrar | girlny, ? |
| 19:24.04 | *** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl) |
| 19:24.12 | Laureano | [TK]D-Fender, which package is incompatible? Libpri or Asterisk? |
| 19:24.16 | carrar | wonders off to do work |
| 19:24.33 | [TK]D-Fender | Laureano: You can't use 1.4 zaptel with 1.2 * |
| 19:25.05 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
| 19:25.38 | Laureano | Mmm, I'm confused now. A few hours ago, someone suggested that I could in #asterisk-dev. |
| 19:25.54 | [TK]D-Fender | Laureano: They are wrong, or you are mistaken |
| 19:26.24 | Laureano | Thanks for the hint. |
| 19:26.31 | Laureano | I'll double check. |
| 19:27.19 | girlny | carrar same server do i have to specify the .sock location in cdr_mysql.conf |
| 19:27.39 | ManxPower-work | Laureano: you really should not be using Asterisk 1.2 if you have ANY choice. |
| 19:27.49 | *** join/#asterisk rossand (n=aross@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com) |
| 19:28.33 | *** join/#asterisk freezey (n=trees@static-64-61-84-174.isp.broadviewnet.net) |
| 19:29.02 | Laureano | ManxPower-work, I know, I know. Long story short: there isn't any chance to upgrade. Really. |
| 19:29.02 | freezey | very odd i have a box that has been running asterisk for quite some time now.. all of the sudden today the proces just started to die starts dies start dies all on its own |
| 19:29.04 | freezey | not sure whats going on |
| 19:29.23 | Laureano | But I use 1.4 and 1.6, when the choice is up to me. |
| 19:29.42 | carrar | girlny, you don't if you connect to it via localhost:3306 |
| 19:29.43 | *** join/#asterisk dkirker-openmobl (n=dkirker@gateway0.openmobl.com) |
| 19:29.59 | ManxPower-work | Laureano: Then you should carefully watch for security related bugs and backport those changes to your system. |
| 19:30.57 | Laureano | ManxPower-work, yes; that's currently one of my duties as consultant for this company. |
| 19:31.15 | *** join/#asterisk ppc (n=ppc@64.72.127.39) |
| 19:32.30 | girlny | carrar the file seen to be fine , what else can i try ? |
| 19:32.44 | carrar | connect to thedb on the command line |
| 19:33.11 | carrar | using what you have in cdr_mysql.conf |
| 19:33.54 | girlny | carrar soory what how do i go in doing that |
| 19:34.24 | carrar | using the mysql command |
| 19:34.52 | girlny | can u show me the first line ? in very new to this |
| 19:35.48 | muiro | A Realtime Queues question: when setting up members in queues.conf, for each member I can set up a Local channel as the channel to call, but then also set an IAX2 channel to use for state information. Can I do this with realtime queue members? I have successfully set it to ring a Local channel, but I don't know how I can set to use the IAX2 channel for state information |
| 19:36.43 | girlny | [root@domU-12-31-39-01-7D-16 ~]# mysql |
| 19:36.43 | girlny | ERROR 1045 (28000): Access denied for user 'root'@'localhost' (using password: NO) |
| 19:37.01 | carrar | mysql -h hostname -p 3306 --user=girlny --password=secret asterisk |
| 19:37.14 | carrar | something LIKE that |
| 19:37.22 | girlny | ok let me tri iy |
| 19:37.23 | carrar | using the values out of your cdr_mysql.conf |
| 19:39.41 | *** join/#asterisk whyz (n=whyz@c-10fce655.1411-32-64736c10.cust.bredbandsbolaget.se) |
| 19:39.57 | carrar | should switch to PostgreSQL |
| 19:39.59 | girlny | carrar i get this mysql Ver 14.12 Distrib 5.0.45, for redhat-linux-gnu (i386) using readline 5. adn alot of liness |
| 19:40.34 | muiro | girlny: that means it worked |
| 19:40.50 | carrar | do you get a prompt? |
| 19:40.51 | carrar | like |
| 19:40.53 | carrar | mysql> |
| 19:41.48 | HenrikJott | and now i have one more question =) when * has completed a call-file sometimes it fails to delete it. The files that * doesn´t delete are empty and with a shorter name than the original. any ideas? |
| 19:41.53 | Katty | well the bill that increases national debit from 12.1 to 12.39 trillion just got passed. |
| 19:42.02 | girlny | no i dont |
| 19:42.14 | girlny | i get a list of values ect |
| 19:42.17 | carrar | so it poped you back to the cli |
| 19:42.19 | carrar | oh |
| 19:42.31 | carrar | so one of the variables is wrong |
| 19:42.38 | titter | Katty: lets celebrate |
| 19:42.41 | carrar | and it exited out of the command with a --help |
| 19:43.37 | muiro | girlny: try mysql -hhostname -ugirlny -psecret asterisk |
| 19:44.08 | whyz | hi. Is there a built-in web interface for asterisk? |
| 19:44.27 | EyeBaw | http://ip:1337/doornob |
| 19:44.32 | carrar | remove the "-p 3306" |
| 19:45.04 | carrar | p should be capital |
| 19:45.52 | muiro | if it's the default port, you don't need to specify it |
| 19:46.27 | muiro | similarly, if the server is on localhost, you don't need the "-h hostname" either |
| 19:46.36 | girlny | so like this mysql -hlocalhost -uadmin -pYngBLb7x6qfb**** asterisk |
| 19:46.53 | muiro | yeah, like that |
| 19:48.07 | girlny | im in |
| 19:48.27 | girlny | i get the mysql> |
| 19:48.32 | freezey | getting a bunch of these asterisk dead but pid file exists |
| 19:49.37 | sbrath | any idea why I would start getting this message from my pri "chan_dahdi.c:12248 dahdi_pri_error: Warning: unknown/inappropriate protocol discriminator received (00/0)" |
| 19:49.40 | muiro | girlny: then your connection works |
| 19:49.51 | sbrath | Been running for a while, and this morning at 3am, that message started. |
| 19:50.19 | sbrath | I'm running libpri-1.4.10.2 |
| 19:51.51 | girlny | ok but when i run domU-12-31-39-01-7D-16*CLI> cdr mysql status |
| 19:51.51 | girlny | Not currently connected to a MySQL server. whats wrong them |
| 19:52.44 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 19:54.06 | [TK]D-Fender | girlny: We aren't psychic and you haven't shown us your configs |
| 19:59.12 | [TK]D-Fender | girlny: Or your even getting the module to load properly |
| 19:59.55 | *** join/#asterisk Corydon76-dig (i=six@c-69-137-80-31.hsd1.tn.comcast.net) |
| 19:59.55 | *** mode/#asterisk [+o Corydon76-dig] by ChanServ |
| 20:00.00 | freezey | really odd that this would happen like this |
| 20:01.28 | *** part/#asterisk acclaimtech (n=kevin@70.62.14.66) |
| 20:02.09 | ManxPower-work | freezey: not odd at all of your asterisk is crashing. "core show uptime" or "show uptime" |
| 20:04.17 | ppc | Whats up |
| 20:04.32 | bmoraca_work | not someones' asterisk, apparently |
| 20:04.37 | freezey | ManxPower-work: just rebooted |
| 20:04.38 | ppc | lul |
| 20:04.49 | freezey | ManxPower-work: check out this error in my /var/log/asterisk/full |
| 20:04.59 | freezey | ManxPower-work: chan_sip.c: Stopping retransmission on '6e4a13163de9fe7441987534249d413f@192.168.30.100' of Request 102: Match Found |
| 20:05.23 | ManxPower-work | freezey: turn off debug |
| 20:06.06 | *** part/#asterisk ManxPower-work (n=EWieling@216.186.151.147) |
| 20:07.01 | bmoraca_work | lol...i love it...my messages log for the last 4 days is 49 bytes |
| 20:08.03 | HenrikJott | exit |
| 20:08.14 | *** part/#asterisk HenrikJott (n=info@d83-183-134-141.cust.tele2.se) |
| 20:08.27 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 20:10.03 | girlny | thaks allot guys you the it workkk problem solveee after lots of hrs |
| 20:11.17 | ppc | So I know the google voice/free phone thing isn't supported etc etc but all of a sudden i'm having problems w/ my server after having the hdd unplugged for a week or two and was wondering if anyone saw any glaring problems in this http://pastebin.com/m6d35336a |
| 20:12.34 | [TK]D-Fender | ppc: unrelated meaningless looking junk... |
| 20:12.54 | ppc | That's what happens when I get an incoming call from GV trying to validate my number |
| 20:13.11 | ppc | My extension isn't rining |
| 20:13.30 | ppc | I can still dial extensions for this iax2 trunk I setup |
| 20:14.03 | *** part/#asterisk |Rain| (i=rain@ev.il.net) |
| 20:14.10 | [TK]D-Fender | ppc: random dialplan. Still meaningless |
| 20:14.54 | ppc | First thing I saw was the "blacklisted" stuff and had no idea what that was |
| 20:17.37 | [TK]D-Fender | ppc: We see it trying to retreive a parked call from a lot we have no proof should even have a call on it. |
| 20:17.38 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:17.38 | *** join/#asterisk toppe_ (n=root@cs181001225.pp.htv.fi) |
| 20:17.39 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:17.39 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 20:23.41 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:24.35 | freezey | seems like out of nowhere my chan_zap.so just started failing out of nowhere |
| 20:29.07 | whyz | this is a bit off topic, but i'd like to know if anyone's got a tip on how to get a simple sip client which doesn't need to do anything but log (or rather trigger a script) when a call comes in. |
| 20:29.31 | whyz | doesn't need to do anything else than react to an incoming call.. not even answer |
| 20:29.46 | bmoraca_work | asterisk can do that |
| 20:30.01 | sbrath | can't you just make a dialplan that does the work and disconnects? |
| 20:30.30 | tzafrir_laptop | freezey, where? |
| 20:30.58 | freezey | my process just keeps dieing each time i try to start it |
| 20:31.08 | freezey | tzafrir_laptop: and that is one of the errors i am noticing in the logs |
| 20:31.12 | whyz | bmoraca_work, ok. i'll take a look.. got any url i could look at? |
| 20:31.24 | bmoraca_work | uhm... asterisk.org ? |
| 20:31.44 | whyz | bmoraca_work, hehe |
| 20:31.46 | tzafrir_laptop | freezey, what do you mean by "fail"? Could you provide the relevant error message? |
| 20:31.48 | whyz | ok thanks anyway |
| 20:32.29 | bmoraca_work | what else did you want? you're not going to find prewritten dialplans to accomplish exactly what you want...and i couldn't show them to you even if they did exist because you haven't told me exactly what you want |
| 20:32.54 | bmoraca_work | asterisk is an application that will perform what you described, and asterisk.org is the homepage for asterisk. |
| 20:33.02 | Katty | pouts. |
| 20:33.06 | freezey | tzafrir_laptop: http://pastebin.ca/1751433 |
| 20:33.12 | Amorsen | whyz: If you don't explicitly Answer() or run one of the few implicitly-answering applications (like Playback()), the call won't get answered |
| 20:35.35 | tzafrir_laptop | chan_zap.c: Unable to open '/dev/zap/channel': No such file or directory |
| 20:35.36 | tzafrir_laptop | ls -l /dev/zap/channel |
| 20:36.18 | whyz | bmoraca_work, i understand that. What i could use is a hint on how to trigger a script on an incoming call. I'm sure that asterisk.org is the place to find such info, but as it's quite a big project and i'm veeery new to both asterisk and sip/voip, i thought i'd save myself some time if i couldbe pointed in the right direction. but it's alright, it'll just take me a bit more time now |
| 20:36.19 | *** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net) |
| 20:36.39 | freezey | i noticed that before.. and on my other asterisk box it has that dir but on the one thats down the dir s gone |
| 20:36.49 | freezey | now i dunnoif the dir was ever there to begin with or not |
| 20:37.07 | bmoraca_work | whyz: look at AGI or "core show application system" |
| 20:38.22 | freezey | tzafrir_laptop: any ideas? |
| 20:39.34 | tzafrir_laptop | freezey, chan_zap.so can't really work if there's no zaptel |
| 20:39.53 | freezey | tzafrir_laptop: i am not sure how the hell it just disappeared though |
| 20:40.21 | tzafrir_laptop | do you have zaptel installed there? |
| 20:41.09 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 20:41.34 | freezey | tzafrir_laptop: was |
| 20:41.35 | *** join/#asterisk lesouvage (n=lesouvag@82.73.69.76) |
| 20:41.39 | freezey | tzafrir_laptop: not sure how it would just disappear |
| 20:42.25 | tzafrir_laptop | is there zaptel hardware there? |
| 20:42.38 | freezey | yes |
| 20:42.41 | freezey | digium cards |
| 20:42.42 | tzafrir_laptop | what kernel version is it? |
| 20:42.44 | tzafrir_laptop | uname -r |
| 20:42.53 | bmoraca_work | freezey, what's lspci say? |
| 20:43.01 | freezey | 2.6.9-55.0.12.ELsmp |
| 20:43.11 | tzafrir_laptop | find /lib/modules -name zaptel.ko |
| 20:43.12 | freezey | 0b:08.0 Ethernet controller: Digium, Inc.: Unknown device 8000 (rev 11) |
| 20:43.41 | freezey | <PROTECTED> |
| 20:44.07 | bmoraca_work | what model card do you have? |
| 20:44.22 | tzafrir_laptop | this is not related to the card. zaptel is not loaded, it seems |
| 20:44.28 | tzafrir_laptop | lsmod | grep zaptel |
| 20:44.32 | tzafrir_laptop | hmm |
| 20:44.35 | tzafrir_laptop | lsmod | grep ^zaptel |
| 20:44.49 | freezey | the one taht is working replys back |
| 20:44.57 | freezey | the one that is dead |
| 20:44.57 | freezey | does not |
| 20:44.58 | bmoraca_work | was going to have him modprobe it again |
| 20:45.13 | tzafrir_laptop | the module zaptel is not loaded |
| 20:45.22 | tzafrir_laptop | let's see what happens if you load the module: |
| 20:45.27 | tzafrir_laptop | modprobe zaptel |
| 20:46.42 | freezey | ok zaptel 186148 0 crc_ccitt 6209 1 zaptel |
| 20:47.21 | tzafrir_laptop | ok, so why wasn't it loaded at startup? |
| 20:47.32 | freezey | not sure why it didnt load |
| 20:47.41 | *** join/#asterisk retentiveboy (n=pdugas@adsl-221-170-128.pns.bellsouth.net) |
| 20:47.43 | tzafrir_laptop | do you have /etc/init.d/zaptel ? |
| 20:47.58 | freezey | yeah |
| 20:49.04 | freezey | ok |
| 20:49.11 | freezey | awesome looked like that fixed it |
| 20:49.12 | freezey | thanks |
| 20:55.40 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
| 20:55.44 | Geminizer | Hello all. |
| 20:56.15 | Geminizer | Does anyone know of a good free international calling service that can be interfaced with asterisk ? |
| 20:56.56 | *** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-170.cablep.bezeqint.net) |
| 20:59.43 | Kobaz | free? |
| 20:59.45 | Kobaz | international? |
| 20:59.51 | Kobaz | those two words don't go together |
| 21:00.14 | Geminizer | That's what I figured :) |
| 21:00.50 | Kobaz | ~itsp |
| 21:00.51 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 21:08.22 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 21:13.51 | *** join/#asterisk minotaur01 (n=minotaur@mail.upperjamestoyota.ca) |
| 21:19.24 | muiro | when Gosub() is used, does it set a channel variable indicating where it got called from in the dialplan? |
| 21:20.07 | Corydon76-dig | No, it uses a channel datastore |
| 21:20.55 | muiro | can I get access to that information from within the gosub? |
| 21:28.15 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-243-188.telstraclear.net) |
| 21:29.21 | *** join/#asterisk xa0z (n=Interex@75-129-244-32.dhcp.mtvr.il.charter.com) |
| 21:29.26 | xa0z | jblack, you alive? |
| 21:29.42 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-243-188.telstraclear.net) |
| 21:31.31 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 21:32.25 | p3nguin | /trout jblack |
| 21:33.05 | sbrath | is their a libpri channel? |
| 21:33.18 | leifmadsen | sbrath: nope |
| 21:33.50 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:33.52 | xa0z | heh |
| 21:34.04 | xa0z | I just wanted to see what he would say p3nguin |
| 21:38.22 | *** part/#asterisk xa0z (n=Interex@75-129-244-32.dhcp.mtvr.il.charter.com) |
| 21:38.59 | *** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net) |
| 21:42.52 | *** join/#asterisk hluesea (n=hulusika@88.239.45.61) |
| 21:47.26 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 21:48.47 | ruben23 | hi if i have this on my dial plan--->exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log) how do i dial on my softphones..? |
| 21:50.48 | *** join/#asterisk phl4kx (n=phl4kx@mail001.centralcorp.upnorte.edu.pe) |
| 21:51.39 | leifmadsen | ruben23: eh? just dial any 2 or more digit extension |
| 21:51.41 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-99-199-10.ph.ph.cox.net) |
| 21:51.55 | leifmadsen | I don't really understand the question |
| 21:53.38 | p3nguin | I think you answered it correctly. |
| 21:56.38 | leifmadsen | seems like a trick question :) |
| 21:56.49 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 21:59.23 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 21:59.49 | *** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194) |
| 22:00.41 | doolittlework | hi there i scratched out an old te110p single span pri card, does anyone know how to get this to work with dahdi? |
| 22:01.29 | [TK]D-Fender | doolittlework: Same as any other similar card |
| 22:02.10 | doolittlework | can't seem to get it to work |
| 22:02.53 | doolittlework | [TK]D-Fender have u received a metal yet |
| 22:03.24 | doolittlework | for you always active and helping out on the channel |
| 22:04.11 | [TK]D-Fender | doolittlework: I own a lot of metal, all "field combat" grade and very sharp |
| 22:04.22 | doolittlework | lol |
| 22:04.45 | carrar | PICS!! |
| 22:06.06 | opticy | am i correct in assuming that the only hardware i need (in addition to a computer w/ a NIC) in order to setup a very simple voip/pstn gateway is a fxo card, such (like this one: http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=160331750263 ) ?? |
| 22:06.49 | [TK]D-Fender | carrar: http://www.roninswords.com/custom_kiku_in_tea.htm |
| 22:07.35 | [TK]D-Fender | carrar: A comission of mine I got a little over a year & a half ago |
| 22:07.50 | [TK]D-Fender | opticy: PSTN is vague. |
| 22:07.56 | [TK]D-Fender | opticy: Depends how. |
| 22:08.15 | *** join/#asterisk cm- (n=chris@sweap.ielf.org) |
| 22:08.18 | [TK]D-Fender | opticy: and the X100 is an ill-advised POS |
| 22:08.26 | carrar | nice |
| 22:08.44 | opticy | lol i figured there had to be some significant inferiorities given the low cost.. |
| 22:08.54 | carrar | a hattori hanso sword ;) |
| 22:09.50 | carrar | looks nice |
| 22:10.14 | opticy | all i'd like to do is be able to access my existing telephone line (which is just a POTS) and get a dialtone from a remotely connected computer |
| 22:11.46 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 22:11.51 | opticy | it'd just be for personal use/tinkering...nothing that would need to support many users |
| 22:13.08 | [TK]D-Fender | carrar: LOL.... no actually its a chinese-made blade from a really good forge (still folded steel, laminated), and differentially tempered |
| 22:13.41 | [TK]D-Fender | carrar: the rest is all custom |
| 22:13.48 | [TK]D-Fender | carrar: Down to the gold work |
| 22:14.23 | carrar | I'll be in Japan again for 2months in a few mons |
| 22:14.25 | carrar | months |
| 22:14.42 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 22:14.42 | carrar | I love going there |
| 22:14.45 | opticy | [TK]D-Fender: what hardware do i need [in addition to a pc/nic] to do this? |
| 22:15.16 | doolittlework | bright spark me disable wrong network card |
| 22:15.19 | [TK]D-Fender | opticy: You want * to use your phone line? |
| 22:15.30 | opticy | yes |
| 22:15.40 | [TK]D-Fender | opticy: Linksys SPA-3102 <- |
| 22:15.41 | opticy | just so i can access my phoneline via voip |
| 22:16.00 | opticy | [TK]D-Fender: that's already pre-built :( |
| 22:16.08 | doolittlework | any one have the lotto numbers i need to retire? |
| 22:16.12 | opticy | i'm more interested in learning about it/building it |
| 22:16.33 | [TK]D-Fender | opticy: huh? |
| 22:17.11 | opticy | i want to use my existing linux box to do it |
| 22:17.17 | [TK]D-Fender | opticy: ..... |
| 22:17.32 | [TK]D-Fender | opticy: Both are jsut devices to use WITH asterisk to give you access to that line |
| 22:17.42 | [TK]D-Fender | opticy: I never said it REPLACED Asterisk |
| 22:18.03 | opticy | perhaps i misread the description...it appeard to be a self-contained router |
| 22:18.26 | [TK]D-Fender | opticy: Yes, its has a lot of other cools stuff on there too... not that you have to USE those aspects |
| 22:18.43 | [TK]D-Fender | opticy: this also lets you use an analog PHONE independent of this with * via SIP as well |
| 22:18.55 | titter | Is there a list of all the reload commands... diaplan reload, iax2 reload, sip reload, etc ... |
| 22:19.05 | [TK]D-Fender | titter: "help" <- |
| 22:20.15 | carrar | module reload! |
| 22:20.56 | titter | was just checking for a queues and meetme reload |
| 22:23.02 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 22:25.13 | *** join/#asterisk catlynx (i=sione@ocs.net) |
| 22:25.33 | catlynx | anyone here uses nexVortex? |
| 22:25.41 | *** join/#asterisk doolittlework (n=d@196.211.34.2) |
| 22:27.24 | doolittlework | [TK]D-Fender do you know what the right driver is for TE110p for dahdi? |
| 22:27.56 | catlynx | must be the only asterisk user with NexVortex |
| 22:29.39 | titter | looks up nexVortex ... webpage 404's |
| 22:29.59 | [TK]D-Fender | doolittlework: doolittlework wcte1XXp |
| 22:31.32 | doolittlework | wcte11xp 28576 0 |
| 22:31.32 | doolittlework | is this the one? |
| 22:32.06 | *** join/#asterisk rossand (n=aross@dhcp-233-179.tb-classrooms.carleton.ca) |
| 22:32.16 | catlynx | titter: my asktreisk refusing to connect with them |
| 22:32.21 | catlynx | err asterisk |
| 22:33.17 | doolittlework | if i do a cat /proc/dahdi/1 i can see my channels but right on top i get IRQ misses: 6096 does anyone know how to fix? |
| 22:33.49 | *** join/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
| 22:33.53 | catlynx | no neat if finaly regstered |
| 22:34.00 | catlynx | brb |
| 22:34.12 | [TK]D-Fender | doolittlework: "cat /proc/interrupts |
| 22:34.57 | doolittlework | 161: 397494 371284 397280 371207 IO-APIC-level uhci_hcd:usb4, wcte11xp |
| 22:35.42 | doolittlework | [TK]D-Fender does this mean my usb 4 is sharing irq with card/ |
| 22:35.44 | doolittlework | ? |
| 22:36.38 | [TK]D-Fender | doolittlework: Yes. Very bad |
| 22:36.58 | doolittlework | must i try to move to other pci slot? |
| 22:38.04 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 22:39.54 | [TK]D-Fender | doolittlework: Or shift assignments around in your BIOS, disable that USB interface, etc |
| 22:40.04 | doolittlework | thx |
| 22:41.37 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 22:43.22 | *** join/#asterisk korihor (n=korihor@201.210.226.98) |
| 22:43.38 | *** join/#asterisk Micc (n=quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 22:43.58 | Chainsaw | doolittlework: PCI roulette usually fixes it though. Depends on how much time you have on your hands. |
| 22:44.30 | *** join/#asterisk Micc (n=quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 22:45.40 | doolittlework | [TK]D-Fender: 169: 28543 26012 28596 25987 IO-APIC-level HDA Intel, wcte11xp i moved it no its clashing with HDA is that on board adio? |
| 22:45.53 | Chainsaw | Correct. |
| 22:46.17 | Chainsaw | You can likely shift the HDA out to MSI though. |
| 22:46.22 | Chainsaw | (Instead of level interrupts) |
| 22:47.04 | doolittlework | if i disabvle onboard sound should clear this errors? |
| 22:47.22 | Chainsaw | If you don't use it, then yes, by all means disable it. |
| 22:47.48 | Chainsaw | Should you want it, continue PCI roulette until you find a slot that's exclusive. |
| 22:47.55 | doolittlework | all looks good now -----169: 22967 27929 22947 27951 IO-APIC-level wcte11xp |
| 22:48.41 | *** join/#asterisk dennis00 (n=dennis@unaffiliated/dennis00) |
| 22:49.04 | doolittlework | strange i still get irq missed on cat /proc/dahdi/1 |
| 22:49.17 | dennis00 | Hi, could people perhaps take a 2nd look to see why my Asterisk incoming calls are not working? http://pastebin.ca/1751631 |
| 22:50.13 | doolittlework | dennis paste somewhere else i can't open link |
| 22:50.49 | Chainsaw | doolittlework: I can. |
| 22:52.06 | doolittlework | go ahead |
| 22:52.16 | dennis00 | http://pastebin.ca/1751631 |
| 22:52.18 | p3nguin | dennis00: This still does not look like a call from OFF-SITE going INTO your * machine. |
| 22:52.32 | dennis00 | p3nguin: is there any chance, the issue is my home router? |
| 22:52.52 | p3nguin | I'm leaning more toward user layer. |
| 22:52.54 | doolittlework | why cant i open the link goes to blank page |
| 22:53.27 | dennis00 | doolittlework: http://pastebin.com/d1f9719e1 does this help? |
| 22:53.34 | p3nguin | dennis00: Why have you never called your DID from a phone that is off of your * system? |
| 22:53.41 | doolittlework | yes |
| 22:54.06 | dennis00 | doolittlework: http://pastebin.com/d1315c4db |
| 22:54.11 | p3nguin | I keep seeing traffic either into your system from the local network or going outbound. |
| 22:54.24 | dennis00 | p3nguin: I did use voipbuster, but I dont have access to a cellphone until Monday :( |
| 22:54.34 | dennis00 | Should I test it and ask somebody to call the nuber? |
| 22:54.37 | p3nguin | dennis00: You don't have a landline phone? |
| 22:54.49 | dennis00 | p3nguin: no, but I will have a cellphone at Monday. |
| 22:55.06 | p3nguin | get someone... anyone... to call the DID while you are in debug mode. |
| 22:55.10 | p3nguin | You want me to call it? |
| 22:55.50 | p3nguin | If possible, make sure there are no other calls in progress when you receive the other call. |
| 22:56.05 | dennis00 | Yes please, ok. |
| 22:56.16 | p3nguin | one moment while I look up the number again. |
| 22:57.02 | doolittlework | dennis00: does yuor incoming come from voipbuster? |
| 22:57.19 | dennis00 | Uh, did you call p3nguin? |
| 22:57.20 | p3nguin | I dialed it, but I don't know if I got it right. |
| 22:57.23 | p3nguin | yes |
| 22:57.23 | doolittlework | o h i see 17.[31107142866] |
| 22:57.40 | p3nguin | Now paste that debug. |
| 22:57.42 | dennis00 | doolittlework: From Voipbuster SIP to my Asterisk SIP |
| 22:58.17 | dennis00 | http://pastebin.ca/1751644 |
| 22:58.20 | p3nguin | I don't speak the language, so I don't know what the lady said to me. |
| 22:58.55 | dennis00 | p3nguin: Did it sound like this? Het persoonlijk nummer 3110... is niet bereikbaar? |
| 22:59.06 | p3nguin | kinda |
| 22:59.20 | dennis00 | lol, great. |
| 22:59.41 | doolittlework | if you using voipbuster as incoming you are pointing calls to context [da-isdn] but you have nothing in there?????? |
| 22:59.45 | p3nguin | I'm still not seeing what I want to see as far as debug goes. |
| 22:59.58 | p3nguin | I thought you were using budgetphone. |
| 23:00.09 | dennis00 | I use budgetphone as incoming, correct. |
| 23:00.17 | dennis00 | doolittlework: see above. |
| 23:00.33 | doolittlework | <dennis00> doolittlework: From Voipbuster SIP to my Asterisk SIP??? |
| 23:00.33 | dennis00 | doolittlework: I use voipbuster to call to budgetphone. Budgetphone is the incoming. |
| 23:00.42 | doolittlework | lol |
| 23:00.56 | dennis00 | Yes. Did I misexplain? |
| 23:01.00 | p3nguin | doolittlework: The problem has been that he is not using an off-site phone to test his DID, so his debug is terrible. |
| 23:01.13 | p3nguin | I dialed it this time, but I still don't see what I want to see. |
| 23:01.42 | p3nguin | Does the box have a public IP address? |
| 23:01.46 | dennis00 | p3nguin: Does it come to looking at extensions.conf to see wht to do with the call? |
| 23:01.50 | dennis00 | p3nguin: Yes. |
| 23:02.06 | p3nguin | No, I just want to see a useful debug. |
| 23:02.11 | doolittlework | add exten => _x.,1,NoOp(incomming from registered sip trunk 31107142866) to inbound context and see id you se the noop when dialing it |
| 23:02.26 | p3nguin | I haven't seen a useful debug all day. |
| 23:02.31 | p3nguin | not one single call INTO the box. |
| 23:02.51 | dennis00 | this box is plain evil. |
| 23:02.56 | p3nguin | Do you want me to login and work on it? |
| 23:03.49 | doolittlework | also check what codex your provider uses i see you have specified no codex in context 31107142866 |
| 23:04.31 | *** join/#asterisk duckz (n=duckz@86.107.84.186) |
| 23:04.46 | dennis00 | p3nguin: that's ok |
| 23:05.00 | dennis00 | doolittlework: like this? [inbound] \n exten => _x.,1,NoOp(incoming from registered sip trunk 31107142866) \n exten => _x,2,Playback(tt-monkeys) |
| 23:05.07 | dennis00 | I should've pastebined that. |
| 23:05.10 | jaskew | I've been following and have a newbie question. Isn;t insecure=very,port redundant? I thought "very" was the same as "invite,port". Also wondering if the "username" directive may be making asterisk expect authentication from budgetphone... |
| 23:05.11 | p3nguin | Without a useful debug, I cannot help you any further. |
| 23:05.21 | doolittlework | yes |
| 23:05.41 | dennis00 | I have sent an e-mail to my ISP, they responded 2 hours ago. |
| 23:05.41 | dennis00 | or VSP. |
| 23:06.24 | dennis00 | they said 1) doolittlework is evil 2) add /31number in register string 3) incoming numbers go to 31number, 4) insecure=very,port |
| 23:06.44 | *** join/#asterisk etfonhomey (n=etfonhom@12.4.217.194) |
| 23:06.55 | p3nguin | That goes to show what they know about asterisk. |
| 23:07.06 | doolittlework | who me? |
| 23:07.49 | dennis00 | no, they said to rmeove the number in the online config as 1), but I dont know what that means, I guess they think i shouldn't post the number on pastebin, but not sure what they mean. |
| 23:08.03 | dennis00 | p3nguin: I have sent you a PB. |
| 23:08.06 | dennis00 | PM* |
| 23:08.16 | p3nguin | Oh, I block PM by default. |
| 23:08.26 | dennis00 | If you wish, you can look in the server. |
| 23:08.32 | p3nguin | Let me unblock you and then you can send again. One moment. |
| 23:08.41 | p3nguin | try again. |
| 23:08.42 | doolittlework | dennis00: incoming numbers go to 31number???????????????????// where do you have this in your inbound context? |
| 23:09.24 | dennis00 | doolittlework: in the register string, ...31107142866@sip1.budgetphone.nl/31107142866, but they explanation isn t very clear. |
| 23:10.02 | dennis00 | i think they mean the /31number |
| 23:11.17 | dennis00 | http://pastebin.com/d17890ed0 < this is what they say. |
| 23:11.45 | doolittlework | s |
| 23:11.56 | doolittlework | 311XXX means any incomming number will go to 3111XXXX you created a sip user with number 311XXX so put it inn your inbound context exten => _311X.,1,DOWHATEVER |
| 23:13.23 | *** join/#asterisk jtodd (i=n36va5gy@asterisk/community-director-and-tie-dye-shirt-lover/jtodd) |
| 23:13.23 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 23:13.25 | dennis00 | I have that in my extensions.conf |
| 23:14.00 | doolittlework | where in [inbound] |
| 23:14.16 | dennis00 | I added it as the 3rd line. |
| 23:14.32 | doolittlework | paste again |
| 23:14.34 | dennis00 | exten => _311X.,1,Playback(tt-monkeys) |
| 23:15.15 | dennis00 | doolittlework: http://pastebin.com/d1baec0fd |
| 23:16.31 | doolittlework | did you load asterisk sounds? |
| 23:16.56 | dennis00 | Yes, it works if I place it under internal and call from asterisk to e.g. 1200 (monkey number) |
| 23:17.38 | doolittlework | you forgot the . in second line _X. |
| 23:18.16 | voipmonk | dennis - you still having issues with that inbound number? :) |
| 23:18.42 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 23:19.05 | dennis00 | voipmonk: Yes, I have been spending the whole day on fixing it. |
| 23:19.21 | dennis00 | But it seems a lot of budgetphone clients had problems fixing it, but it works. |
| 23:19.58 | doolittlework | dennis http://pastebin.com/m5a5d8257 |
| 23:20.50 | dennis00 | thank you. |
| 23:21.03 | dennis00 | p3nguin is currently logged in, let's hope he can find something. |
| 23:21.06 | p3nguin | dennis00: fixed |
| 23:21.16 | p3nguin | I got the monkey recording. |
| 23:21.18 | dennis00 | really?? |
| 23:21.25 | dennis00 | OMG |
| 23:21.26 | p3nguin | Now where do you want the call to go? |
| 23:21.28 | p3nguin | 1001? |
| 23:21.29 | dennis00 | that' s sooooooo great!!! |
| 23:21.33 | dennis00 | Yes please. |
| 23:21.38 | dennis00 | THANKS MAN!!! |
| 23:22.13 | doolittlework | dennis paste extensions.conf |
| 23:22.57 | voipmonk | he matched _311X. ? |
| 23:23.01 | dennis00 | http://pastebin.com/m3c590075 |
| 23:23.05 | dennis00 | We've got monkeyssss! |
| 23:23.20 | dennis00 | I think it's the first. |
| 23:23.27 | dennis00 | But I am not sure. |
| 23:23.51 | voipmonk | well find out - and add it to your notes |
| 23:25.09 | doolittlework | <doolittlework> 311XXX means any incomming number will go to 3111XXXX you created a sip user with number 311XXX so put it inn your inbound context exten => _311X.,1,DOWHATEVER |
| 23:25.09 | *** join/#asterisk Godfather_ (n=Godfathe@79.109.251.13.dyn.user.ono.com) |
| 23:25.30 | doolittlework | dennis you need to make yourselve familiar on how the context and extensions work |
| 23:25.36 | voipmonk | no - that means any number starting with 311 |
| 23:25.48 | dennis00 | I know, I have gotten very confused. |
| 23:26.45 | doolittlework | what do you mean no? |
| 23:26.54 | doolittlework | voipmonk |
| 23:27.19 | *** join/#asterisk dennis00 (n=dennis@unaffiliated/dennis00) |
| 23:27.27 | dennis00 | that is the 3rd time today, but I'm back. |
| 23:27.32 | doolittlework | lol |
| 23:27.49 | doolittlework | my turn |
| 23:27.56 | doolittlework | for help |
| 23:28.00 | doolittlework | please |
| 23:28.07 | doolittlework | te110p |
| 23:28.55 | doolittlework | keep on getting IRQ missed if i do cat /proc/dahdi/1 |
| 23:30.33 | doolittlework | if i only get 7 misses is that still badd |
| 23:31.09 | doolittlework | use to be 4000 when it was sharing irq's with sound card |
| 23:33.13 | doolittlework | k noone whats to hel me maybe i must change my nick to wannadowork |
| 23:34.09 | *** join/#asterisk uluatu (n=uluatu@189.114.176.219.dynamic.adsl.gvt.net.br) |
| 23:35.17 | *** join/#asterisk uluatu_ (n=uluatu@200.146.5.244.dynamic.dialup.gvt.net.br) |
| 23:36.57 | doolittlework | http://pastebin.com/m1cabfa68 if anyone can point me in the right direction |
| 23:38.48 | *** join/#asterisk uluatu (n=uluatu@189.114.176.219.dynamic.adsl.gvt.net.br) |
| 23:39.58 | p3nguin | Wow, it barely costs me more to call dennis00 than it does to call someone in my same city. |
| 23:40.23 | dennis00 | Yes, it's very cheap to a normal number. |
| 23:40.52 | bmoraca_work | what locality? |
| 23:41.42 | dennis00 | The Netherlands |
| 23:41.46 | bmoraca_work | ahh |
| 23:41.51 | bmoraca_work | nowhere that matters :P |
| 23:41.52 | dennis00 | only calling to mobiles is expensive. |
| 23:42.56 | Nugget | Ik ben een vliegende koe. |
| 23:43.16 | p3nguin | I call within the USA, 1.05 cents/min... I called him in Netherlands, 1.93 cents/min. |
| 23:43.49 | bmoraca_work | 1.05 cents/min? that's nearly twice my average rate to the US |
| 23:44.27 | p3nguin | What provider are you using? |
| 23:48.49 | leifmadsen | 1.1c/m is a pretty average rate for non-high volume users (i.e. small business/residential) |
| 23:48.59 | leifmadsen | to get better rates you usually have to buy a lot of minutes |
| 23:50.20 | dennis00 | 1.1c/m, I want that. |
| 23:50.25 | dennis00 | I even pay more to the Netherlands lol |
| 23:50.27 | dennis00 | and I live there. |
| 23:50.48 | *** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net) |
| 23:50.48 | dennis00 | it's free when I pay 9.50euro monthly, but I could take a landline if I wnated to pay 120euro yearly. |
| 23:51.37 | dennis00 | Nugget: You are not a flying cow. |
| 23:53.12 | *** join/#asterisk bcnyc (n=bill@pool-71-187-81-196.nwrknj.east.verizon.net) |
| 23:53.56 | Nugget | I am too! |
| 23:54.11 | *** part/#asterisk profxavier (n=chatzill@unaffiliated/neverblue) |
| 23:54.13 | Nugget | Boe Boe! |
| 23:54.21 | *** join/#asterisk TheDavidFactor (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 23:59.14 | *** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be) |
| 23:59.42 | dennis00 | Hey! That's what Dutch cows say |