| 00:01.06 | *** join/#asterisk PepOSX (n=pepOSX@190.72.148.91) |
| 00:01.34 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 00:05.01 | *** join/#asterisk BBHoss (n=hoss@146.229.191.76) |
| 00:07.59 | *** join/#asterisk alephcom (n=chatzill@h66-112-187-16.mcsnet.ca) |
| 00:10.07 | ta^3 | I'm unable to success use a TC400B, even using file convert works. firmware and module are succesfully loaded and detected by asterisk (transcoder show) |
| 00:10.46 | ta^3 | I'm trying to convert/bridge between IAX2/G711u and SIP/G.729a. |
| 00:11.33 | mosty | what error do you get? |
| 00:12.01 | ta^3 | mosty: no errors, just does not works. |
| 00:12.33 | ta^3 | I mean, the files created (or the call) is no audio. |
| 00:13.07 | ManxPower | ta^3: does "show translations" show that G729 is active? |
| 00:13.18 | ta^3 | ManxPower: yes, it is. |
| 00:13.41 | ManxPower | ta^3: what is your allow= and disallow= lines in sip.conf? |
| 00:13.55 | mosty | ta^3, so there's nothing in the full log at all? |
| 00:13.58 | ta^3 | disallow=all, allow=g729 |
| 00:14.14 | ManxPower | ta^3: I recommend you contact Digium support. |
| 00:15.06 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 00:15.26 | ta^3 | mosty: nothing useful nor related. |
| 00:15.44 | ManxPower | Generally only Digium support can help with G729 issues |
| 00:16.17 | mosty | do you have codec_g729a.so installed at the same time? |
| 00:16.22 | ta^3 | I should wait for tomorrow morning, in order to call. |
| 00:16.53 | ta^3 | mosty: there is no codec_g729a.s |
| 00:16.54 | JT | TC400B, isn't that a TestCase400Board? ;) |
| 00:17.03 | ta^3 | JT: seems to. ;-) |
| 00:17.26 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 00:19.19 | ManxPower | mosty: you should not need codec_g729a if you have the hardware transcoder board. -- which is why I said that he should Call Digium Support |
| 00:19.30 | *** join/#asterisk macros73_ (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 00:19.48 | mosty | ManxPower, i know, but i think it's worth making sure it's not present |
| 00:22.17 | mamep | anyone can help me with oooh323 channel? |
| 00:22.19 | mamep | http://pastebin.ca/781810 |
| 00:24.44 | BBHoss | wow that TC400B board is expensive |
| 00:25.26 | ManxPower | BBHoss: That is to be expected. |
| 00:25.35 | ta^3 | mosty: it isn't. BBHoss, that means that it should work :) |
| 00:26.01 | mosty | BBHoss, you have to factor in the cost of the g729 licences |
| 00:26.24 | BBHoss | oh so you dont have to pay licensing fees |
| 00:26.42 | ta^3 | BBHoss: well i think they are included in the price. |
| 00:26.43 | ManxPower | BBHoss: the licensing fees are built into the cost of the board |
| 00:26.58 | BBHoss | licensing alone would be $960, not including g723.1 |
| 00:27.12 | ManxPower | BBHoss: you can't know what the licensing would be. |
| 00:27.37 | mosty | you can compare to the cost of that many software licences |
| 00:27.38 | ta^3 | also consider the non-cpu-power used. |
| 00:27.44 | BBHoss | mamep: h323 support blows on *, i would suggest using chan_woomera and hook it up to OPAL |
| 00:28.06 | BBHoss | they ought to make one that does speex encoding |
| 00:28.30 | ManxPower | mosty: not really. If you compared the cost to the same number of software licenses you would have to factor in the cost of having several more machines to run them |
| 00:28.31 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-e4efe4ed22ee0f3d) |
| 00:28.31 | BBHoss | or an other non-patent encumbered product |
| 00:29.01 | mosty | ManxPower, obviously, but at least it gives you some idea |
| 00:30.42 | ManxPower | The only reason to use G729 and G723.1 is for interoperating with products that do not support any highly compressed codec other than G729 and G723.1 |
| 00:31.22 | mosty | fujin, allow=preferred_codec does not work, asterisk complains that preferred_codec is an unknown format |
| 00:31.33 | hmmhesays | haha I just watched a video of peter frampton playing "black hole sun" |
| 00:31.33 | BBHoss | ManxPower: so it doesn't reduce cpu usage? |
| 00:32.06 | mamep | chan_woomer? |
| 00:32.07 | BBHoss | it might also decrease the time needed to transcode from speex to ulaw |
| 00:32.08 | mamep | chan_woomera? |
| 00:32.23 | mamep | supports user/pass based authentication? |
| 00:32.24 | BBHoss | yeah, anthm wrote it a while back |
| 00:32.39 | BBHoss | OPAL handles that |
| 00:33.05 | mamep | and what's opal? |
| 00:33.24 | BBHoss | an h323/SIP server |
| 00:33.49 | ManxPower | mosty: please put down the crack pipe and step away from the computer. |
| 00:33.58 | BBHoss | OPAL is like a kind of gatekeeper, whereas Woomera links OPAL into asterisk |
| 00:33.59 | bkw_ | chan_woomera doesn't work with OPAL Woomera |
| 00:34.07 | bkw_ | the protocol has changed a little bit |
| 00:34.11 | mosty | ManxPower, i was just trying something that fujin said would work |
| 00:34.36 | mamep | well i want to connect to h323 cisco and route my calls through h323 |
| 00:34.39 | mamep | is is possible? |
| 00:34.41 | bkw_ | yes |
| 00:34.47 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929633.dsl.bell.ca) |
| 00:34.52 | BBHoss | bkw_: when did this happen |
| 00:34.56 | ManxPower | mosty: then you are a moron, as any non-moron would realize that "allow=preferred_codec" means "allow=ulaw" |
| 00:35.04 | bkw_ | BBHoss: when woomera was done in OPAL |
| 00:35.12 | bkw_ | OpenH323 woomera the older version works with chan_woomera |
| 00:35.14 | BBHoss | hmm too bad |
| 00:35.27 | bkw_ | but to do OPAL woomera the channel driver will need an update |
| 00:35.46 | mamep | where can i find some guidance? |
| 00:35.49 | ManxPower | bkw_: So, as usual H323 support is screwed up. |
| 00:35.59 | BBHoss | heh |
| 00:36.08 | bkw_ | ManxPower: no just use the older H323/Woomera |
| 00:36.11 | bkw_ | and not the OPAL version |
| 00:36.15 | BBHoss | why is h323 so neglected |
| 00:36.15 | mosty | ManxPower, well then that's a very bad way of phrasing it, since it does not do what i said i was trying to do |
| 00:36.21 | ManxPower | You would think that with something like FIVE different H323 drivers for Asterisk, one of them would work well. |
| 00:36.46 | bkw_ | BBHoss: we have the start of Mod_opan in FreeSWITCH |
| 00:36.48 | ManxPower | mosty: Did you tell fujin what codec you prefer to use? |
| 00:36.55 | bkw_ | er mod_opal |
| 00:37.10 | mamep | anyone? |
| 00:37.19 | mosty | ManxPower, yes. i prefer to use whichever is the caller's highest preference that i support |
| 00:37.30 | ManxPower | mosty: you can't do that with Asterisk |
| 00:37.37 | *** join/#asterisk MooingLemur (n=troy@unaffiliated/mooinglemur) |
| 00:37.58 | ManxPower | mamep: almost everyone that tries to use H323 with Asterisk gives up. |
| 00:38.07 | BBHoss | only problem is that OpenH323 apparently isnt being maintained anymore |
| 00:38.27 | ManxPower | mosty: Asterisk has no support for figuring out what codec the client prefers. |
| 00:38.57 | mosty | ManxPower, iax has codecpriority=caller - what does that do then? |
| 00:38.59 | bkw_ | BBHoss: i'm good friends with Craig |
| 00:39.10 | bkw_ | mosty: that means we side with the caller |
| 00:39.12 | bkw_ | anthm did that patch |
| 00:39.18 | ManxPower | mosty: I thought we were talking about SIP |
| 00:39.25 | mamep | ManxPower : and what's your suggestion? |
| 00:39.34 | ManxPower | mamep: find a way to use SIP. |
| 00:39.35 | mosty | manxpower: i am trying to find the equivalent for sip |
| 00:39.37 | bkw_ | BBHoss: OPAL is the replacement for OpenH323 |
| 00:39.54 | ManxPower | mosty: do you have any indication that chan_sip supports that feature? |
| 00:39.54 | mamep | bkw_ : where can i find opal? |
| 00:39.57 | BBHoss | yeah |
| 00:40.01 | mamep | ManxPower : no way |
| 00:40.02 | bkw_ | www.opalvoip.org |
| 00:40.06 | bkw_ | www.woomeravoip.org |
| 00:40.19 | mosty | ManxPower, that is what i'm trying to figure out |
| 00:40.23 | ManxPower | mamep: then expect to spend several weeks trying to get it to work, and don't expect and real help from people on this channel., |
| 00:40.29 | ManxPower | mosty: I already told you. |
| 00:40.39 | bkw_ | if I can get things lined out we'll have mod_opal done before the first FreeSWITCH release |
| 00:41.02 | mamep | thx |
| 00:41.19 | mosty | ok, so it's not possible with sip on asterisk, then i can stop working on this |
| 00:42.30 | BBHoss | OPAL is going to be integrated in Afelio when we get a release |
| 00:42.51 | bkw_ | Afelio you mean if you get a release? I wasn't aware any code was written yet |
| 00:43.15 | ManxPower | mosty: chan_iax gets many features before chan_sip does. qualify smoothing, and jitter buffer are both things chan_iax had years before chan_sip. |
| 00:43.15 | BiG^DoG | hey, if I'm getting a lot of dropped calls on my PSTN line, what log file would I check? /var/log/asterisk/full? |
| 00:43.18 | BBHoss | yeah its just not in the repos |
| 00:43.45 | bkw_ | BBHoss: I just don't really get the whole Afelio stance |
| 00:43.48 | ManxPower | BiG^DoG: set callprogress=no and busydetect=no in /etc/asterisk/zapata.conf Those are just aliases for randomlydropmycalls= |
| 00:44.00 | bkw_ | BBHoss: I guess once I see code then i'll understand and see the direciton |
| 00:44.01 | bkw_ | er direction |
| 00:44.05 | mosty | ManxPower, as long as i can say to my boss "it's not my fault, asterisk doesn't support that" then i am safe, heh |
| 00:44.22 | BBHoss | yeah its taken a while for me to fully understand |
| 00:44.30 | BiG^DoG | ManxPower: sarcasm? |
| 00:44.41 | ManxPower | BiG^DoG: only the last part. |
| 00:44.45 | bkw_ | BBHoss: I still don't get what was so wrong with FreeSWITCH :P |
| 00:46.31 | ManxPower | BiG^DoG: the most common cause of dropped calls is having callprogress or busydetect set to yes. |
| 00:46.46 | BiG^DoG | but I'm not using a zaptel interface... I'm using a SPA3102 |
| 00:47.10 | ManxPower | BiG^DoG: next time mention that in your original question. |
| 00:47.21 | BiG^DoG | my original question was what log file to check |
| 00:47.25 | ManxPower | BiG^DoG: I have no suiggestions now that I learned. |
| 00:47.26 | mamep | i have to connect to callmanager or gw? |
| 00:47.51 | ManxPower | BiG^DoG: check whatever is specified in /etc/asterisk/logger.conf |
| 00:48.42 | ManxPower | but very seldom will you see the cause of a dropped call in the logs, unless it's a reinvite or similar NAT related issue, then you will get mas retransmittions /retrys messahges |
| 00:49.51 | BiG^DoG | I am getting lots of "stopping retransmission" errors |
| 00:51.05 | BiG^DoG | is that a symptom of my problem? |
| 00:53.26 | BBHoss | anybody ever used a Zapmicro prodcut? |
| 00:55.17 | ManxPower | BiG^DoG: it COULD be. That would indicate a NAT, firewall, or reinvite issue. Try setting canreinvite=no in sip.conf |
| 00:55.43 | *** part/#asterisk snazm (n=snazm@78.147.13.67) |
| 00:55.49 | BiG^DoG | is that set per extension I assume |
| 00:56.53 | ManxPower | BiG^DoG: no. per device listed in sip.conf. extensions are listed in extensions.conf |
| 00:57.03 | BiG^DoG | sorry |
| 00:57.07 | BiG^DoG | poor word choice |
| 00:57.20 | ManxPower | canreinvite= MIGHT be supported in sip.conf [general] but I would have to look at sip.conf.sample to know for sure. |
| 00:58.53 | BiG^DoG | doesn't matter... didn't work... set canreinvite=no and nat=no and still getting |
| 00:58.53 | ManxPower | BiG^DoG: in the telecom world, one wrong word can cost you tens of thousands of dollars. |
| 00:58.53 | BiG^DoG | sorry! :) |
| 00:58.54 | BiG^DoG | Nov 18 19:58:20 DEBUG[11315] chan_sip.c: Stopping retransmission on '647d6ff070a382f941e9f17161581f8a@192.168.1.71' of Request 102: Match Found |
| 00:59.13 | ManxPower | BiG^DoG: the match found means it is not the cause of your issue. |
| 01:01.22 | *** join/#asterisk salzh (n=salzh@124.77.5.180) |
| 01:02.19 | *** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com) |
| 01:04.57 | ManxPower | turning on DEBUG will let you see many messages that look bad, but are normal |
| 01:08.29 | BiG^DoG | yeah .. and this is a stupid trixbox server... I'm so thinking about rebuilding it as a simple, plain * box... Too many things can change my .conf files without my knowledge ... a plain ol' * server will do what I tell it |
| 01:11.35 | ManxPower | ~zeeek |
| 01:11.45 | jbot | from memory, zeeek is someone who once said "learning asterisk using a GUI is like learning sex through masturbation. You'll never get to the good stuff." |
| 01:13.45 | hmmhesays | LOL |
| 01:13.47 | hmmhesays | rock on |
| 01:15.54 | blitzrage | rock lobster |
| 01:17.31 | [TK]D-Fender | blitzrage, know what he leading cause of death in lobsters is? |
| 01:17.41 | Corydon76-dig | Boiling? |
| 01:17.42 | blitzrage | rocking out too hard? |
| 01:17.45 | [TK]D-Fender | blitzrage, Festivals :) |
| 01:18.01 | blitzrage | <clap></clap> |
| 01:18.18 | BiG^DoG | the wife is being surprisingly supportive of all these "screwed up" phone calls lately so I'm gonna keep pushing it! :) |
| 01:19.12 | blitzrage | thank goodness I don't have to deal with a wife |
| 01:19.22 | blitzrage | I can do anything I want :) |
| 01:19.28 | Corydon76-dig | or a husband |
| 01:19.29 | *** join/#asterisk kclaussen (n=kclausse@204.13.224.242) |
| 01:19.34 | blitzrage | or that |
| 01:19.40 | blitzrage | Corydon76-dig: your FB status changed... ? |
| 01:19.57 | Corydon76-dig | blitzrage: yes, I'm single now |
| 01:20.03 | blitzrage | pure crazyness |
| 01:20.09 | blitzrage | welcome to the club |
| 01:20.39 | Corydon76-dig | So do you no longer feel safe? ;-) |
| 01:21.49 | blitzrage | I never felt safe :) |
| 01:22.25 | blitzrage | wow |
| 01:22.38 | hmmhesays | being recently single sucks |
| 01:22.42 | hmmhesays | but after that its ok |
| 01:22.56 | ManxPower | blitzrage: wait until you see my take on christmas 8-) |
| 01:23.35 | Corydon76-dig | "Happy Pagan Winterfest"? |
| 01:23.35 | blitzrage | hmmhesays: ya... going from lots of sex to no sex sucks... but other that that... the space is nice :) |
| 01:23.35 | blitzrage | Happy Festivus |
| 01:23.38 | Corydon76-dig | blitzrage: I can help you with the no sex part |
| 01:23.43 | blitzrage | ManxPower: I don't like holidays in general |
| 01:23.53 | blitzrage | Corydon76-dig: actually, you can't -- you have the wrong parts |
| 01:24.23 | Corydon76-dig | blitzrage: How can anything that feels so right be wrong? |
| 01:24.26 | ManxPower | blitzrage: I think that depends on which side you are looking at. |
| 01:24.51 | blitzrage | ummm.... I guess? |
| 01:24.58 | [hC] | is there any way to query the current time as displayed on a polycom (without having physical access to the phone of course) |
| 01:25.35 | blitzrage | well, I'm off to finish my detox on the couch for the rest of the night! |
| 01:25.43 | Corydon76-dig | Heh |
| 01:25.46 | [hC] | detox? what th... oh i guess it is sunday |
| 01:25.47 | [hC] | :) |
| 01:29.33 | fujin_ | Corydon76-dig: that's a little too homosexual for most |
| 01:29.58 | mosty | [hC], do the polycom's have a log page on their website? on snom phones i just look at their last log entry for a rought idea if the time is correct |
| 01:30.01 | Corydon76-dig | fujin_: really? ;-) |
| 01:30.32 | [hC] | mosty: yeah i can look at the log.. it looks right, but it seems weird that the time would have fixed itself on its own since i last looked on friday.. i dunno.. |
| 01:32.39 | mosty | [hC], to a packet trace and watch for ntp traffic |
| 01:33.51 | *** join/#asterisk dlynes (n=dlynes@d154-20-45-103.bchsia.telus.net) |
| 01:50.32 | mosty | fujin_, for most homophobes i guess yeah haha |
| 01:50.55 | fujin_ | Not really. |
| 01:51.07 | fujin_ | Just non-homosexuals. |
| 01:51.18 | fujin_ | erm, what's the word? Hetero |
| 01:53.32 | JT | hetro isn't it |
| 01:53.44 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 01:54.28 | fujin_ | perhaps |
| 01:54.47 | fujin_ | mosty: are you implying that disliking the thought of a homosexual blowjob makes me a homophobe? |
| 01:54.53 | fujin_ | by definition, that'd also mean I'm afraid of the thought |
| 01:55.17 | mosty | not quite |
| 01:55.24 | JT | fujin_: well, it's not coincidental that homophobes are usually always the ones that feel the need to blurt out comments like their dislike of certain acts |
| 01:55.31 | JT | fujin_: subtle homophobia perhaps |
| 01:55.32 | mosty | but maybe |
| 01:56.00 | fujin_ | see, I'm not afraid, I just dislike. |
| 01:56.07 | fujin_ | People tell me this means I'm a homophobe. |
| 01:56.18 | fujin_ | by definition, this does not compute |
| 01:56.21 | JT | fujin_: yes basically it's about not being rude |
| 01:56.28 | fujin_ | was I? |
| 01:56.37 | Corydon76-dig | How can you dislike that which you've never experienced? |
| 01:57.04 | fujin_ | Now, there's a question. |
| 01:57.07 | JT | fujin_: yes |
| 01:57.42 | fujin_ | Face value of the thought, for what it's worth. |
| 01:57.47 | fujin_ | It's not sexually appealing to me, at all. |
| 01:57.51 | fujin_ | That's how. |
| 01:57.55 | JT | fujin_: there is no need to tell anyone |
| 01:58.00 | JT | it's subtle homophobia |
| 01:58.06 | fujin_ | I'm sure, not disagreeing |
| 01:58.58 | fujin_ | So, what's the opposite of homophobia? |
| 01:59.02 | JT | it's like saying "eww, black people having sex, how disgusting" |
| 01:59.12 | Corydon76-dig | fujin_: homophile |
| 01:59.24 | Corydon76-dig | or homophilia |
| 01:59.29 | fujin_ | And what does that define? |
| 01:59.36 | fujin_ | Liking homosexual people? |
| 01:59.40 | fujin_ | or, not being afraid? |
| 01:59.41 | bkw_ | you like them too much? |
| 01:59.51 | bkw_ | doesn't that mean you're gay? |
| 01:59.52 | bkw_ | : |
| 01:59.52 | bkw_ | p |
| 02:00.20 | fujin_ | I'm just not sure why homophile/homophiliacs can blurt out "homophobe" whenever a hetro makes a statement. |
| 02:00.21 | Corydon76-dig | It might, but I know women who are homophiles while not being homosexual |
| 02:00.34 | fujin_ | That's very true. But women are women, that's a moot point |
| 02:01.13 | Corydon76-dig | I've also met some very open minded straight men who tried it, just to be sure |
| 02:01.15 | JT | fujin_: only when it's a homophobic statement |
| 02:01.29 | *** join/#asterisk shido6 (i=shido6@74-130-126-198.dhcp.insightbb.com) |
| 02:01.50 | JT | there's a lot to be said for keeping ones mouth closed if what comes out will be a prejudiced comment |
| 02:01.51 | fujin_ | sorry, missed the earlier question, does homophile/philia define "liking homosexual people", or "not being afraid of homosexual people"? |
| 02:02.40 | mosty | homophilia just means love of the same |
| 02:03.08 | fujin_ | I see. |
| 02:04.53 | fujin_ | Don't get me wrong, I didn't mean to offend or seem prejudiced. I'm not a homophile, nor homophobe. |
| 02:08.27 | *** part/#asterisk bkw_ (n=brian@adsl-64-149-41-151.dsl.tul2ok.sbcglobal.net) |
| 02:28.46 | lowlevel | mmm tassimo |
| 02:29.19 | *** join/#asterisk mtaht4 (n=m@125-105-62-200.enitel.net.ni) |
| 02:31.24 | rob0 | This may sound weird, but to me, to find out that someone's homosexual is a major factor in their favor. The reason being that homosexuals endure much adversity and hatred, and that sort of thing tends to build character. |
| 02:31.38 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.pa.comcast.net) |
| 02:33.26 | *** part/#asterisk mtaht4 (n=m@125-105-62-200.enitel.net.ni) |
| 02:33.39 | [TK]D-Fender | rob0, Or perhaps it merely mimics stereotypes of BDSM loving masochists ;) |
| 02:33.50 | rob0 | :) |
| 02:43.02 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 02:50.21 | *** join/#asterisk mackes (n=root@cpe-76-180-236-119.buffalo.res.rr.com) |
| 02:50.21 | mosty | hmm, i just re-routed about 15 simultaneous g729 calls from one asterisk server to another, and the load dropped from about 30 down to 5. the machine has dual xeon 3.6Ghz cpu's, why would such a small change in g729 usage create a large drop in system load? |
| 02:51.27 | mackes | g729 requires transcoding |
| 02:52.25 | fujin_ | mosty: what codec to<>from? |
| 02:52.30 | fujin_ | is it all g729<>g729? |
| 02:52.33 | mackes | that conversion from Ulaw to 729 takes some horsepower, especially with Asterisk |
| 02:52.41 | mackes | It must be staying in the call |
| 02:52.54 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-89-254.hag.east.verizon.net) |
| 02:54.31 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 02:55.58 | mosty | on this server, i did have about 40 simultaneous g729 calls, and the load ranged from 30 to 50. then i took some of those clients and made them register to another server, that took about 12 g729 calls off the original server, and the load dropped to under 5 |
| 02:57.00 | *** join/#asterisk TheDamn3d (n=hidden@bas2-quebec09-1242405607.dsl.bell.ca) |
| 02:57.07 | [TK]D-Fender | mosty, because it was passing the call off "as-is" without decoding the audio. |
| 02:57.16 | *** part/#asterisk TheDamn3d (n=hidden@bas2-quebec09-1242405607.dsl.bell.ca) |
| 02:58.09 | mosty | i thought 12 g729 transcoded calls would take less system load than that |
| 02:59.53 | [hC] | mosty: depends what kind of server it is, but the load should not have been that high. unless by load you mean cpu usage percentage? |
| 03:00.14 | [hC] | oh i see the spec up there. |
| 03:00.36 | [hC] | i transcode 30+ channels on single xeon 2.4ghz machines and the cpu load only hits about 50% |
| 03:00.38 | mosty | perhaps i'm not using the optimal g729 module on that box, what should i be using for a 32bit dual xeon 3.6G? |
| 03:00.52 | [hC] | i686 module |
| 03:01.20 | mosty | cpu load was very high. i've done more g729 calls in the past on this box, with a much lower cpu load. i'm not sure what has changed |
| 03:01.20 | *** join/#asterisk salmander (n=salmande@CPE0002b323839e-CM001225024984.cpe.net.cable.rogers.com) |
| 03:01.38 | salmander | Does anyone know if IAX is trademarked? |
| 03:01.57 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.itb.ac.id) |
| 03:10.56 | mosty | salmander, why? |
| 03:12.19 | salmander | picking a name for a service need to know if I can use IAX as part of the name, do you know? |
| 03:13.57 | mosty | teliax use it as part of their name |
| 03:14.17 | salmander | ok, would you know where i |
| 03:14.23 | salmander | er, where I could find out for sure? |
| 03:14.49 | mosty | ask digium, or else i'm sure there's a registry that you can search in your state/country |
| 03:15.45 | *** join/#asterisk Op3r (n=edwin@203.177.221.73) |
| 03:17.02 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-37dd8364e044cc35) |
| 03:20.34 | mackes | Have any of you worked with FreeSwitch? |
| 03:20.40 | mackes | or OpenSER? |
| 03:21.08 | salmander | *wanted to look at freeswitch, but another time-consuming project poped up* |
| 03:21.39 | mackes | yeah. |
| 03:22.01 | mackes | I was thinking about putting OpenSER or FreeSwitch in front of my asterisk installation |
| 03:24.51 | *** join/#asterisk smurf (n=smurf@debian/developer/smurf) |
| 03:29.12 | *** join/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 03:29.35 | Zuchmir | how healthy is a 40,000+ line conf file for * ? |
| 03:29.44 | mackes | not |
| 03:30.44 | mackes | I'm sorry, I'm not comparing the two... I believe the freeswitch scales... I am attempting to understand who it is for? |
| 03:31.02 | mackes | Only ITSP? |
| 03:31.21 | mackes | roadrunner, Verizon, FWD, SipPhone? |
| 03:34.37 | *** join/#asterisk `Sauron (i=sauron@dsl001-130-033.aus1.dsl.speakeasy.net) |
| 03:38.02 | [TK]D-Fender | Zuchmir, my worst didn't pass 500...... and Iw as trying :) |
| 03:41.46 | *** join/#asterisk bmg505 (n=leon@196.209.177.134) |
| 03:44.41 | Zuchmir | [TK]D-Fender: we have approx 10000 files, in multi-level menu system |
| 03:45.14 | Zuchmir | [TK]D-Fender: public lectures which we make available |
| 03:48.09 | linagee | does asterisk use ICMP when you do sip to sip calls? |
| 03:48.15 | linagee | some sort of making sure things are alive? |
| 03:51.22 | [TK]D-Fender | Zuchmir, I sincerely doubt you though about how to abstract that properly. |
| 03:52.00 | [TK]D-Fender | linagee, No, you have rtp timeout if not reinviting, and qualify |
| 03:52.30 | linagee | [TK]D-Fender: here is the tethereal dump. 216 asterisk friend is trying to call me. http://pastebin.com/m1bbe5cdd |
| 03:52.51 | linagee | [TK]D-Fender: "ICMP Destination unreachable (Host administratively prohibited)" |
| 03:53.12 | JT | linagee: that's just a standard ICMP message |
| 03:53.19 | [TK]D-Fender | linagee, pleawse provide SIP debug... from * |
| 03:53.29 | linagee | [TK]D-Fender: we get ringing, but no audio |
| 03:53.32 | linagee | [TK]D-Fender: sec |
| 03:54.08 | JT | linagee: if the host is firewalled, then yes, this is a problem |
| 03:55.18 | linagee | JT: hrm |
| 03:56.40 | Zuchmir | [TK]D-Fender: you are correct, however step 1 in implementation has to match old system 1-to-1 and as such the old structure with all it's problems must be adhered to |
| 03:57.13 | [TK]D-Fender | Zuchmir, I still can't buy that you couldn't have shunk it considerably and still meet spec. |
| 03:57.52 | JunK-Y | mooo |
| 03:59.06 | [TK]D-Fender | JunK-Y, I'm had a whole lot of moo these past weeks :) |
| 04:00.16 | JunK-Y | heres a new one: moooooooo ;) |
| 04:12.55 | linagee | should rtpchecksums = yes or not? |
| 04:12.57 | linagee | no |
| 04:13.07 | *** join/#asterisk groogs[h] (n=gregm@cbl-66-102-80-229.wtccommunications.ca) |
| 04:13.50 | JT | linagee: if you are getting the ICMP message you said you got from one of your hosts on the SIP call, you have problems outside of asterisk |
| 04:14.13 | linagee | JT: i had a firewall problem on my side |
| 04:14.26 | linagee | JT: i had some "icmp prohibited" message blocked |
| 04:14.40 | linagee | JT: strange part is i unblocked that. now i can hear him but he still can't hear me |
| 04:14.52 | JT | linagee: ok |
| 04:15.24 | linagee | JT: and we are getting some sort of [UDP CHECKSUM INCORRECT] errors |
| 04:15.35 | JT | from what? |
| 04:15.58 | *** join/#asterisk ManxPower (n=manxpowe@232.sub-70-197-184.myvzw.com) |
| 04:16.07 | linagee | JT: tethereal |
| 04:16.20 | linagee | JT: sec. we are both setting it to rtpchecksums=yes |
| 04:16.42 | *** join/#asterisk acidfu (n=acidfu@modemcable176.199-56-74.mc.videotron.ca) |
| 04:17.49 | ManxPower | linagee: why would you set that to yes? |
| 04:18.59 | linagee | ManxPower: *shrug* |
| 04:19.04 | linagee | ManxPower: just trying to get it to work |
| 04:19.24 | linagee | so that's strange. when i ring him, i can hear him but he can't hear me. when he rings me, we both can't hear each other. (but it rings) |
| 04:19.44 | linagee | SIP is sooooo fun. :-( |
| 04:19.58 | JT | linagee: yeah perhaps give us a summary of your setup |
| 04:20.08 | linagee | JT: i am using freepbx, he is not |
| 04:20.19 | linagee | JT: i've had success calling and being called by another freepbx user |
| 04:20.35 | JT | wrong channel? ;) |
| 04:20.39 | linagee | :P |
| 04:20.43 | ManxPower | linagee: almost all one-way audio issues are nat, firewall, or bindaddr issues. |
| 04:20.52 | linagee | JT: he has SIP phones that connect to him over the internet which is weird |
| 04:21.03 | JT | what's weird about that? |
| 04:21.16 | JT | and btw, that's not really a description of the scenario at all |
| 04:21.19 | linagee | ManxPower: it's using amazon EC2 so i don't doubt it's a firewall issue. :( |
| 04:21.26 | fujin_ | http://www.craigslist.org/about/best/nyc/51760058.html |
| 04:21.32 | asdx | this is very weird, i can hear sound from my softphone (zoiper) when i do test with ael-demo, etc, but when i call someone through my voip provider (teliax) i can't hear the other person, but he does receive/accept my calls |
| 04:21.38 | asdx | http://pastebin.com/m3271e24e |
| 04:21.42 | asdx | that's the output when i dial |
| 04:21.54 | ManxPower | linagee: set canteinvite=no in sip.conf |
| 04:21.58 | JT | asdx: sounds like a standard nat related misconfiguration |
| 04:22.01 | JT | again, not weird ;) |
| 04:22.07 | JT | ~sipnat |
| 04:22.08 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 04:22.16 | asdx | JT: it was working before, and i'm using iax2 |
| 04:22.23 | asdx | JT: nat should be irrelevant with iax2 right? |
| 04:22.39 | JT | asdx: not necessarily |
| 04:22.44 | JT | iax2 is just downright unreliable |
| 04:22.52 | JT | i try to avoid it outside of softphone testing |
| 04:22.58 | ManxPower | JT: It took me all of 10 mins to make NAT + SIP + Asterisk work, then another hour or so to make the SIP ATA I was using work without reconfiguraiton as it moved from behind nat, to not begind nat. |
| 04:23.05 | ManxPower | what is so complicated about that |
| 04:23.09 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.139.96) |
| 04:25.37 | [TK]D-Fender | asdx, [Nov 19 01:19:18] WARNING[18817]: channel.c:3012 set_format: Unable to find a codec translation path from g729 to gsm <-- what part of "gee I guess you don't have G.729 licenses" is not blatantly obvious to you here? |
| 04:31.21 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 04:34.59 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 04:35.50 | asdx | [TK]D-Fender: licenses? |
| 04:36.29 | [TK]D-Fender | asdx, G.729 is a paid codec for * and must be ordered through Digium's site |
| 04:37.05 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 04:37.12 | obnauticus | For holdmusic, what are the specifications do I need? |
| 04:37.16 | obnauticus | like mono, stereo |
| 04:37.19 | obnauticus | 16bit |
| 04:37.20 | obnauticus | 8bit |
| 04:37.22 | obnauticus | what? |
| 04:37.30 | asdx | [TK]D-Fender: but i could talk before, what do you think is using G.729? |
| 04:37.50 | ManxPower | asdx: because the error message SAID G729 |
| 04:38.04 | [TK]D-Fender | asdx, Do you need a BIGGER giant flashing neon sign? |
| 04:38.27 | asdx | [TK]D-Fender: no |
| 04:40.12 | asdx | i understand that i need the codec |
| 04:40.35 | asdx | but i don't understand why i could talk before with this person that i'm calling now, well... it worked with another provider |
| 04:40.41 | asdx | not with teliax |
| 04:40.48 | mosty | does "show g729" say you have used all your licences? |
| 04:41.03 | *** join/#asterisk Buhntz (i=Boones@port-212-202-42-179.dynamic.qsc.de) |
| 04:41.31 | [TK]D-Fender | asdx, because odds are you set your codec improperly when you switched to using IAX2 and never looked |
| 04:41.40 | ManxPower | asdx: when you do something like "allow=all" you will have random horroble codec issues. don't use allow=all use disallow=all and then allow= for the codec you want. |
| 04:42.28 | asdx | ManxPower: i have disallow=all allow=gsm right now |
| 04:43.18 | ManxPower | asdx: not for one leg of the call. |
| 04:43.27 | ManxPower | all calls have two legs |
| 04:43.43 | ManxPower | one leg is trying to use gsm, one leg is trying to use g729 |
| 04:50.43 | asdx | yeah |
| 04:51.17 | obnauticus | [TK]D-Fender, Asterisk will re-encode my on hold music to whatever codec I have set for that preticular channel, correct? |
| 04:51.40 | obnauticus | automatically to whatever i want as long as the player can read it? |
| 04:51.50 | [TK]D-Fender | obnauticus, if ti can, yes |
| 04:51.54 | obnauticus | k |
| 04:51.56 | obnauticus | just making sure. |
| 04:52.48 | asdx | so how can i find what leg is trying to use g729 |
| 04:53.35 | mosty | asdx, how many g729 licences are free/in use when you do "show g729" ? |
| 04:54.36 | [TK]D-Fender | sakjdhjkasdlkhksadh |
| 04:54.38 | [TK]D-Fender | omg.... |
| 04:54.49 | obnauticus | sak? |
| 04:55.25 | [TK]D-Fender | just so sad... |
| 04:55.28 | ManxPower | asdx: Ok, so you are using a SIP phone to call thru asterisk to where? |
| 04:55.42 | asdx | ManxPower: i'm using iax |
| 04:55.48 | asdx | mosty: "no such command" |
| 04:55.48 | [TK]D-Fender | ManxPower, its BEYOND blatantly obvious in his pastebin. |
| 04:55.54 | [TK]D-Fender | ManxPower, http://pastebin.com/m3271e24e |
| 04:56.13 | ManxPower | asdx: OK, so exactly what device to you personally speak into when trying to call this person? |
| 04:56.45 | [TK]D-Fender | asdx, Get a set of eyes. Call comes in GSM, you dial Teliax (have to even LOOKED at your peer config?!?!) and Teliax says... G.729 please! |
| 04:56.57 | [TK]D-Fender | ManxPower, Zoiper on the server itself |
| 04:57.32 | ManxPower | [TK]D-Fender: OH. he's beyond my help then |
| 04:57.59 | asdx | [TK]D-Fender: i see |
| 04:58.00 | [TK]D-Fender | ManxPower, Seriously. I suggest extreme voltage. |
| 04:59.04 | asdx | i'm very new to this sorry |
| 04:59.22 | asdx | i'm trying to learn |
| 04:59.23 | ManxPower | [TK]D-Fender: I can think of only one thing. The dreaded /ignore. |
| 05:00.03 | ManxPower | asdx: first, do not try to run a voip softphone on the same machine as your asterisk server. It massively complicates things. |
| 05:00.18 | [TK]D-Fender | asdx, this qualifies for the "z0mg can't see the giant flashing neon sign". |
| 05:00.45 | [TK]D-Fender | ManxPower, but is completely unrelaetd to the fact he still hasn't gotten a clue and started loking at his >>>>>>>>>IAX2 PEER<<<<<<< |
| 05:00.57 | [TK]D-Fender | asdx, ******HINT******* |
| 05:01.08 | asdx | [TK]D-Fender: i seen the message before, i was just wondering why it wanted that codec, since i been using teliax for some time and never seen that message before... |
| 05:01.41 | [TK]D-Fender | asdx, who says IT wanted that codec? the only reason they ACCEPTED with G.729 is because you are OFFERING IT. |
| 05:02.10 | ManxPower | asdx: zoiper is trying to use that codec or teliax is trying to use that codec, You can stand here and argue all night (I don't care), but it is not going to work until you fix it. |
| 05:02.54 | ManxPower | now. fix your iax.conf and make sure you are not doing a direct dial in the dialplan |
| 05:03.38 | ManxPower | [TK]D-Fender: the more I think about, the more I bet he's dialing by IP in Asterisk, hence bypassing iax.conf |
| 05:03.56 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 05:04.12 | [TK]D-Fender | ManxPower, Executing [999@default:1] Dial("IAX2/100-2", "IAX2/puli2007@teliax/01161405378955") in new stack |
| 05:04.22 | [TK]D-Fender | ManxPower, Not unless he modded his hosts file for it :) |
| 05:04.36 | [TK]D-Fender | ManxPower, we;ve got ourselves another cut & paste coder. |
| 05:04.42 | ManxPower | [TK]D-Fender: don't tempt fate. I could not pull up his pastebin. |
| 05:04.44 | [TK]D-Fender | ManxPower, With 0 clue. |
| 05:04.52 | [TK]D-Fender | ManxPower, expired? |
| 05:05.10 | [TK]D-Fender | ManxPower, Here, I refreshed it for you :) http://pastebin.com/m407a29a9 |
| 05:05.44 | ManxPower | -- Unregistered IAX2 '100' (UNAUTHENTICATED) |
| 05:06.47 | ManxPower | asdx: disallow=all and allow=gsm in iax.conf [general] and under each other [whatever] sections of iax.conf, issue an "reload" in the ASterisk CLI |
| 05:06.48 | asdx | http://pastebin.com/macc329a |
| 05:06.54 | ManxPower | until you do that, we cannot help yo |
| 05:07.01 | asdx | that's my config |
| 05:07.12 | [TK]D-Fender | asdx, SET. YOUR. CODECS. |
| 05:07.19 | [TK]D-Fender | asdx, you've vonfigured next to nothing. |
| 05:07.31 | ManxPower | asdx: I do not see any disallow= or allow= lines in the [teliax] section. |
| 05:07.42 | asdx | oops |
| 05:07.49 | ManxPower | why do I not see any disallow= or allow= lines in the [teliax] section? |
| 05:08.00 | asdx | i thought the disallow/allow stuff was only for the softphone.... |
| 05:08.03 | ManxPower | And while you are at it put them in the [general] section now. |
| 05:08.19 | asdx | ManxPower: ok |
| 05:08.20 | ManxPower | asdx: there is nothing really different between teliax and a softphone |
| 05:09.57 | ManxPower | asdx: I SAID earlier that all calls have two legs (in your case the softphone and teliax) and that each leg should have disallow=all and allow=gsm. I don't care how new you are at this, I can't imagine any way to be any more clear. |
| 05:10.38 | *** join/#asterisk chendy (n=chendy@121.76.132.123) |
| 05:10.43 | obnauticus | asdx think of it like a bigass route to the caller :P |
| 05:10.44 | [TK]D-Fender | asdx, thats like asking for someone to send you an e-mail and NOT telling them that you only speak 1 language and they decide " Hey, yeah! Swahili ought to do just fine! .... and Swahili just isn't it!" |
| 05:11.10 | asdx | so if i set disallow=all allow=gsm in [general] that will apply to all the context or whatever is called right? |
| 05:11.19 | asdx | it will make global? |
| 05:11.45 | asdx | i see the point now |
| 05:11.46 | asdx | thx :) |
| 05:12.03 | asdx | i didn't specify the codecs in [teliax] |
| 05:12.29 | [TK]D-Fender | asdx, the go DO IT |
| 05:13.40 | asdx | http://pastebin.com/m332dafef |
| 05:13.40 | asdx | done |
| 05:13.40 | ManxPower | [TK]D-Fender: I still think you, me, JT, and maybe one or two others should just take a break from #asterisk for a while |
| 05:14.32 | asdx | i hope i didn't bother with my n00b questions |
| 05:14.55 | [TK]D-Fender | asdx, Se it in you PEER and stop relying on GLOBAL crap. |
| 05:15.39 | JT | ManxPower: going batty? |
| 05:16.21 | asdx | [TK]D-Fender: peer is [teliax] [100] etc? |
| 05:16.31 | [TK]D-Fender | asdx, BOTH |
| 05:16.37 | asdx | [TK]D-Fender: ok, thank you |
| 05:16.48 | *** join/#asterisk booray (n=ray@64.70.85.2) |
| 05:17.14 | asdx | http://pastebin.com/m79d7fb93 |
| 05:17.50 | ManxPower | JT: Mostly just curious what would happen. |
| 05:18.30 | booray | I want to run Asterisk in a linux Virtual Machine with server 2003 as the host OS. Is there a way to make a TDM400P work with this setup? I don't think the VM can have direct access to the PCI bus... |
| 05:18.55 | fujin | no, and that's a dumb idea |
| 05:18.58 | booray | haha |
| 05:19.10 | booray | well it works great in a VM when it's just voip |
| 05:19.10 | JT | ManxPower: heh |
| 05:19.51 | booray | fujin: why? |
| 05:20.07 | [TK]D-Fender | booray, Sure, try under load and wher you need timing :) |
| 05:20.56 | booray | [TK]D-Fender: I'm not talking any crappy workstation server... assuming the hardware was strong enough to handle the latency, is such a thing even possible? |
| 05:21.14 | fujin | no, sharing a pci device is not |
| 05:21.15 | fujin | afaik |
| 05:21.21 | fujin | maybe in Xen, I've not had much experience with it. |
| 05:21.30 | [TK]D-Fender | .... |
| 05:21.34 | [TK]D-Fender | ~wglwat |
| 05:21.35 | jbot | i heard wglwat is well, good luck with all that |
| 05:21.39 | fujin | indeed |
| 05:21.45 | fujin | I wouldn't do it |
| 05:21.49 | fujin | hell, I wouldn't use a TDM400P. |
| 05:22.18 | booray | is there even a w32 driver that could turn a tdm400p into 4 com ports? i.e. then throw the vm four serial ports and fool zaptel into something or other |
| 05:22.29 | booray | fujin: this one was free. :) |
| 05:22.29 | fujin | DONT DO IT |
| 05:22.32 | fujin | FOR THE LOVE OF GOD |
| 05:22.33 | booray | haha |
| 05:22.45 | [TK]D-Fender | booray, You clealy have no clue about this hardware... |
| 05:22.45 | fujin | Install Linux on your 'strong hardware' |
| 05:22.52 | fujin | run a win32 server in a vm if you must |
| 05:23.04 | [TK]D-Fender | booray, And Win32 drivers for Digium cards.... LOL |
| 05:23.07 | booray | I'm a masochist, so what? |
| 05:23.13 | fujin | you're an idiot, so what |
| 05:23.22 | booray | I wouldn't go that far |
| 05:23.27 | booray | I haven't done it yet, have I? |
| 05:23.30 | [TK]D-Fender | booray, .... |
| 05:23.33 | [TK]D-Fender | ~wglwat |
| 05:23.34 | jbot | hmm... wglwat is well, good luck with all that |
| 05:23.34 | tzafrir_home | booray, 4 com ports? what good is that for? |
| 05:23.34 | fujin | no, but you're thinking of doing it |
| 05:23.45 | [TK]D-Fender | booray, You clearly have hardware and time to waste. have fun! |
| 05:23.47 | booray | That's why I am here, to ask questions, etc, brainstorm |
| 05:23.48 | tzafrir_home | there are dirt-cheap cards that can do that |
| 05:24.19 | fujin | tzafrir: he wants to make the tdm400p into something that vmware (or equivalent) can virtualise to a guest |
| 05:24.21 | booray | it's all theory at this point anyway |
| 05:24.27 | fujin | wow, vmware access to a PCI device.. that's just wrong |
| 05:24.34 | [TK]D-Fender | tzafrir_home, No, he's thinking of about virtualizing Digium cards :) |
| 05:25.08 | tzafrir_home | I think it basically works with Xen on a Linux host |
| 05:25.20 | fujin | ^^. |
| 05:25.24 | linagee | do jitterbuffers only exist for iax? or can you use jitterbuffer with sip too? |
| 05:25.28 | fujin | It's probably not worth doing it, time vs. money. |
| 05:25.44 | tzafrir_home | With qemu on a linux host it works, but I'm not sure about the performance |
| 05:26.36 | booray | So you'd just assume spend the extra money and put asterisk on it's own physical box on it's own special footprint than try and share it with additional hardware |
| 05:27.00 | *** join/#asterisk SwK (n=SwK@user-24-214-55-149.knology.net) |
| 05:27.09 | tzafrir_home | booray, make that system e.g. a Xen host |
| 05:27.17 | tzafrir_home | and run windows on it as well |
| 05:27.27 | fujin | ~cheap |
| 05:27.28 | jbot | from memory, cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 05:27.35 | fujin | Asterisk is *very* hardware sensitive. |
| 05:27.38 | tzafrir_home | Though if you plan a decent load on that system, you should consider a dedicated system |
| 05:28.13 | booray | Look, I'm trying to not buy *two* multi-thousand dollar servers |
| 05:28.22 | fujin | that's fine |
| 05:28.29 | fujin | why not? |
| 05:28.32 | tzafrir_home | booray, this should work well on CPUs that support native virtuallization . e.g. all Intel CoreDuo II |
| 05:28.39 | luke-jr | booray: so don't run Windows |
| 05:28.46 | fujin | s/CoreDuo.*/Xeon/ |
| 05:28.55 | booray | tzafrir_home: thanks, I'm looking into the Xen thing, just pulled up the site |
| 05:29.20 | fujin | booray: virtualise asterisk, get some dedicated hardware to terminate your lines into SIP |
| 05:29.22 | tzafrir_home | fujin, surely not all Xeons. "Xeon" is a brand Intel has used for over 10 years |
| 05:29.23 | fujin | problem solved |
| 05:29.41 | tzafrir_home | booray, latest Linux distros support it quite natively |
| 05:29.52 | booray | fujin: ata style? |
| 05:29.59 | *** join/#asterisk mihinomenest (i=GjI0@66.255.220.17) |
| 05:30.00 | fujin | No. Uh, |
| 05:30.01 | SwK | booray, how many lines you have? |
| 05:30.04 | fujin | I'm talking pri<>SIP |
| 05:30.07 | fujin | E1/T1 etc. |
| 05:30.12 | booray | luke-jr: I'm not even gonna argue. :) |
| 05:30.19 | fujin | I use a Cisco AS5400, with two e1 cards |
| 05:30.21 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 05:30.29 | fujin | dual powersupplies, dual uplinks, redundant swithing fabric |
| 05:30.30 | booray | I'll give it to you short. let me type, one second |
| 05:30.46 | SwK | AS5300 for <= 4 T1/E1s are super cheap these days |
| 05:30.52 | fujin | ^^. |
| 05:30.58 | fujin | THe as5400 has ben very handy |
| 05:31.02 | SwK | yeah |
| 05:31.08 | fujin | Saves alot of hastle. |
| 05:31.11 | tzafrir_home | Yeah, but then again, it's SIP and not real TDM |
| 05:31.13 | SwK | 5400s are a bit more expensive then the 5300... |
| 05:31.22 | tzafrir_home | some things don't really work wel over it |
| 05:31.27 | fujin | that's true |
| 05:31.32 | fujin | (i.e.; fax) |
| 05:31.32 | SwK | like what? |
| 05:31.34 | SwK | wrong |
| 05:31.37 | fujin | no? |
| 05:31.41 | SwK | fax works great w/ AS gateways |
| 05:31.46 | SwK | its called T38 or T37 |
| 05:31.46 | fujin | I've not tried. |
| 05:32.05 | luke-jr | fax works fine w/ ulaw ☺ |
| 05:32.19 | fujin | ulaw<>sip<>ulaw->Fax? |
| 05:32.21 | SwK | works fine w/ ulaw on a LAN |
| 05:32.24 | tzafrir_home | SwK, when have you last used T38 with Asterisk? |
| 05:32.27 | luke-jr | fujin: yep |
| 05:32.29 | luke-jr | do it all the time |
| 05:32.35 | tzafrir_home | When have you seen hardware implementing T37? |
| 05:32.36 | luke-jr | actually, IAX2 not SIP |
| 05:32.40 | SwK | tzafrir, who said anything about allowing asterisk to handle by faxes :P |
| 05:32.40 | booray | A business voip provider would be nice, but the DSL available doesn't allow for decent latency if any bandwidth is in use, QoS or not. A T1 loop appears the only way to get a decent service level agreement with any voip provider. That's $400/month for data only. I have a free TDM400P card and some extra time to put into it, so I thought why not just do 2-4 analog lines for now? It seems to be the most cost effective. I can set it up quickly as I have severa |
| 05:33.14 | booray | So going with a pri style anything just seems a little overkill for my current needs |
| 05:33.21 | tzafrir_home | booray, makes sense |
| 05:33.29 | SwK | tzafrir_home, ALL cisco AS gateways support T37... and once you have it to t37 pretty much any moron w/ a copy of ${MTA} can use it |
| 05:33.40 | tzafrir_home | PRI is indeed more fun to work with, but analog will do for now |
| 05:34.37 | SwK | doesnt take a rocket scientist to figure out that T37 is pretty much just SMTP w/ the fax attached as a tiff |
| 05:34.48 | luke-jr | lol |
| 05:34.52 | fujin | That's handy |
| 05:34.56 | fujin | why'd the choose tiff, I wonder |
| 05:35.16 | SwK | fujin, thats the format faxes are transmitted in :P |
| 05:35.18 | luke-jr | no problems bitmap has? |
| 05:35.22 | fujin | oh, I see. |
| 05:35.25 | fujin | didn't know that. |
| 05:35.33 | luke-jr | SwK: faxes aren't raw data? |
| 05:35.37 | tzafrir_home | booray, start with a spare machine. Not a 1000$ / 2000$ one. Just your spare old server. Stick the card into it |
| 05:35.40 | SwK | luke-jr, nope... |
| 05:35.45 | fujin | I would've assumed fax was raw, yeas. |
| 05:35.49 | mosty | tiff supports multiple pages |
| 05:35.55 | tzafrir_home | How many concurrent calls do you expect? |
| 05:36.04 | luke-jr | mosty: sortof |
| 05:36.10 | booray | tzafrir: thanks. I'll either do that or investigate a multi-line ata or something, as I think that would be good too |
| 05:36.21 | SwK | not its 1Bit tiff-g3 mostly.. |
| 05:36.32 | luke-jr | SwK: colour faxes? |
| 05:36.37 | SwK | actually tiff does support multiple pages |
| 05:36.55 | fujin | booray: a couple of pap2t's analogue<>SIP |
| 05:37.06 | SwK | luke-jr, TIFF supports colour... there's a specific spec for colour faxes but I cant remember which image encoding spec it uses |
| 05:37.11 | tzafrir_home | If you're thinking of up to 10 concurrent calls or so, even with compressed codec transcoding, then this is not too big a task for every P3 computer |
| 05:37.32 | luke-jr | anyone know how to hack PAP2 firmware? |
| 05:37.46 | mosty | hassle your provider |
| 05:38.18 | SwK | if you are only going to have a few POTS lines you can use either TDM400 (or the like) or if you want a hardware gateway look at something more like a AudioCodes MediaPack Gateway like a MP114 or MP118 |
| 05:38.18 | luke-jr | I have no provider |
| 05:38.19 | luke-jr | it's -NA |
| 05:38.27 | mosty | then why do you need to hack it? |
| 05:38.33 | SwK | luke-jr, yeah what mosty just said |
| 05:38.34 | luke-jr | I want to add IPv6 support |
| 05:38.41 | SwK | hah |
| 05:38.43 | SwK | call cisco |
| 05:38.49 | luke-jr | :/ |
| 05:39.12 | SwK | or get one of those cheap ATAs that have the OSS Firmware and hack ipv6 into that |
| 05:39.22 | luke-jr | bleh |
| 05:39.26 | mosty | booray, actually, get a sangoma card with hardware echo cancellation instead of a tdm400 |
| 05:39.30 | luke-jr | SwK: no such thing? |
| 05:40.54 | SwK | luke-jr, i think there is such a thing |
| 05:41.19 | tzafrir_home | mosty, he already has hte card |
| 05:41.39 | luke-jr | SwK: not that I've seen |
| 05:41.43 | luke-jr | or else I'd have got it originally |
| 05:42.01 | tzafrir_home | For a small 4 ports system, a software echo can will do in most cases |
| 05:43.53 | tzafrir_home | What's the state of ipv6 support in Asterisk? |
| 05:44.15 | tzafrir_home | Still requires a special branch, right? |
| 05:44.23 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 05:45.20 | *** join/#asterisk GuyOCanada (i=GuyOCana@75.155.220.205) |
| 05:45.46 | GuyOCanada | Hello |
| 05:46.25 | GuyOCanada | I have created a dialplan for my inbound sip trunk (Is there a way to test it locally without the need of dialing my DID number)? |
| 05:47.06 | [TK]D-Fender | GuyOCanada, What are you expecting to get from this "test"? |
| 05:47.43 | GuyOCanada | Well I want to make sure that all my IVR menus and stuff work |
| 05:48.09 | [TK]D-Fender | GuyOCanada, make another exten your phone can dial nad just goto your IVR. |
| 05:49.46 | TJNII | Any opinions on the Lynksys SPA941? |
| 05:50.23 | TJNII | ~SPA941 |
| 05:50.24 | jbot | from memory, spa941 is an affordable, feature rich IP business phone shown here: http://www.zingotel.com/online/en/business/SPA941?PHPSESSID=6603853f4081dffad6966eab01b162a7 |
| 05:50.34 | TJNII | Awesome |
| 05:50.39 | TJNII | Christmas present +1 |
| 05:50.48 | [TK]D-Fender | TJNII, Where are you located? |
| 05:51.05 | TJNII | Iowa |
| 05:51.32 | [TK]D-Fender | TJNII, Linksys is OK, but Polycom is in the same ballpark and a better phone. |
| 05:51.35 | GuyOCanada | huh? |
| 05:51.39 | [TK]D-Fender | www.telephonydepot.com |
| 05:51.57 | *** join/#asterisk ManxPower (n=manxpowe@232.sub-70-197-184.myvzw.com) |
| 05:52.05 | [TK]D-Fender | GuyOCanada, Just make an exten that leads to you menu for testing. |
| 05:52.09 | *** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net) |
| 05:52.29 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 05:53.08 | GuyOCanada | using GOTO? |
| 05:53.19 | TJNII | [TK]D-Fender: Would you say a Soundpoint 301 is comperable? |
| 05:53.35 | [TK]D-Fender | TJNII, forget the IP 301 unless you have a killer deal. |
| 05:53.46 | TJNII | $114? |
| 05:53.53 | [TK]D-Fender | TJNII, If you have a port to spare on your switch, get an IP 320 + power brick |
| 05:53.58 | [TK]D-Fender | TJNII, no good. |
| 05:54.19 | [TK]D-Fender | IP 301 = char matrix display, no speakerphone. |
| 05:54.30 | TJNII | Oooh, I want speakerphone |
| 05:54.34 | GuyOCanada | [TK]D-Fender: can you give me an example please |
| 05:54.34 | [TK]D-Fender | otherwise a great phone, but the IP320/330 are better choices. |
| 05:55.02 | [TK]D-Fender | GuyOCanada, Exten => 123,1,Goto(contextwithmyivr,whateverextenitson,1) |
| 05:55.30 | TJNII | Oh, there's the 320 on page 4 |
| 05:56.50 | GuyOCanada | so if i want to go to sip-incoming exten => s,1,Playback(welcome) i would do Goto(sip-incoming,s,1)? |
| 05:57.11 | [hC] | i am really liking the 320/330s |
| 05:57.15 | [hC] | the new standup style is great. |
| 05:57.42 | [TK]D-Fender | GuyOCanada, yes |
| 05:58.12 | [TK]D-Fender | [hC], IP320/330 is really hard to beat for typical enterprise use. |
| 05:58.23 | [hC] | you said it. |
| 05:58.39 | TJNII | [hC]: I'm thinking about giving it to my Dad. I'm planning on setting up a * box for them for christmas since VoIP will save them money and give them features my Mom's buisness could really use. Would you recommend it as a gift? Would someone not familar with feature ritch phones figure it out? |
| 05:58.59 | [hC] | the only thing that could compare depending on what sort of a design pickiness you have is an aastra 480i/51i |
| 05:59.15 | [hC] | TJNII: yes. its a great phone for the average user. |
| 05:59.30 | [TK]D-Fender | [hC], Actually the Linksys is pretty decent. I'd probably prefer it over the Aastra... |
| 05:59.45 | mosty | i prefer snom phones to linksys |
| 05:59.46 | [TK]D-Fender | [hC], I don't find the Aastra's handling very nice. |
| 05:59.58 | TJNII | Good. I'm looking for remote-configuration, good speakerphone, good call quality, and caller ID with name. |
| 06:00.04 | [hC] | [TK]D-Fender: the spa9xx ? im not sure about that.. it would be a close call. I say aastra because of how you can flexibly extend it.. it depends on the circumstance |
| 06:00.05 | mosty | the only downside of the snom's is the snom headsets are worthless |
| 06:00.18 | [hC] | i really gotta try the new snom line |
| 06:00.24 | TJNII | I'll be there for initial set up, but afterwords it really needs to provision off the server. |
| 06:00.48 | [TK]D-Fender | [hC], Yeah the Aastra is WAY more configurable, and that IS nice, but I might choose to do without for the more comfortabl call handling and that applies to the button usage as well. |
| 06:00.59 | JT | way prefer linksys to snom |
| 06:01.01 | [hC] | [TK]D-Fender: yep. agreed. |
| 06:01.07 | JT | snom are incredibly ugly |
| 06:01.11 | [TK]D-Fender | [hC], In N/A Polycom just destroys Snom. |
| 06:01.11 | mosty | snom's are easy to autoprovision, polycom aren't much more trouble. but the snom web interface is much better, if you need to login remotely to change settings |
| 06:01.12 | JT | and the lcd screen design is godawful |
| 06:01.31 | [hC] | snom recenly changed their design enough to become more attractive to me |
| 06:01.35 | mosty | JT: i find the linksys phones ugly, personally |
| 06:01.45 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 06:01.52 | [TK]D-Fender | mosty, That I would certainly think is reasonable given the few screenshots I've seen of it |
| 06:01.52 | mosty | the snom's with the large lcd look ok to me |
| 06:03.25 | [TK]D-Fender | mosty, Thing is for $200USD the screen my be large, but it looks like crap. |
| 06:03.28 | *** join/#asterisk grimsy (n=chatzill@203-206-220-214.perm.iinet.net.au) |
| 06:03.47 | [TK]D-Fender | ~phones |
| 06:03.47 | jbot | from memory, phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. ... |
| 06:03.57 | mosty | for my use, i never really needed a phone with a large lcd anyway |
| 06:04.34 | [hC] | heh |
| 06:04.40 | [hC] | aastra above cisco? hell no. |
| 06:04.47 | [TK]D-Fender | mosty, Sonm 360 = 128x64BL 3"x1.75". Polycom IP 501 = 4" x 2" 160x180G |
| 06:04.52 | [hC] | again i guess it depends what you're using to rank that. |
| 06:05.13 | *** join/#asterisk l0verb0y (n=l0verb0y@210.1.137.41) |
| 06:05.16 | mosty | [TK]D-Fender, what do you use the lcd for? callerid? |
| 06:05.27 | *** part/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-37dd8364e044cc35) |
| 06:05.34 | [TK]D-Fender | [hC], Above Cisco because of cost, the fact Aastra support presence and has usable soft-keys and better call handling & PoS |
| 06:05.36 | [TK]D-Fender | PoE* |
| 06:05.40 | dlynes | mosty: Aastra 57i and Aastra 480i has an even larger LCD |
| 06:05.44 | [hC] | yeah. |
| 06:05.50 | [TK]D-Fender | mosty, And Microbrowser, etc |
| 06:05.53 | dlynes | mosty: you can use LCD on those phones for interactive menus, ... |
| 06:06.03 | mosty | menus of what? |
| 06:06.10 | [TK]D-Fender | dlynes, Downside of Aastra LCD = char martrix! |
| 06:06.21 | dlynes | [TK]D-Fender: char matrix? |
| 06:06.22 | [TK]D-Fender | mosty, XHTML services of course |
| 06:06.24 | [hC] | [TK]D-Fender: have you heard of anything about polycom 601's being eol'ed in favor of the 650? i cant seem to get ahold of any 601's from my distributor any more, bastards.. |
| 06:06.28 | *** join/#asterisk DaveCanoe (n=Dave@canoe404.dclg.ca) |
| 06:06.31 | [TK]D-Fender | dlynes, Character matrix, not pixel based. |
| 06:06.41 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-2abed33790964e0c) |
| 06:06.43 | JT | mosty: i have a snom 360 here |
| 06:06.48 | mosty | [TK]D-Fender, what kind of services? |
| 06:06.48 | dlynes | [TK]D-Fender: since when? |
| 06:06.48 | [TK]D-Fender | [hC], Nope, haven't heard that before, but it is believable. |
| 06:06.57 | JT | mosty: huge screen, worst design ever, looks terribl, ugly as all hell :) |
| 06:06.57 | [TK]D-Fender | dlynes, Sice AFAICT always. |
| 06:07.15 | dlynes | [TK]D-Fender: the 57i can do graphics...don't know about the 480i |
| 06:07.31 | JT | i think the snom is char matrix |
| 06:07.40 | l0verb0y | hey |
| 06:07.44 | [TK]D-Fender | dlynes, while the boot logo is graphical, look at the RUN-TIME. I ran a 57i as my primary phone for a few months |
| 06:07.50 | [TK]D-Fender | dlynes, HATED it. |
| 06:07.51 | mosty | JT, i know some people hate the way they look. i'm not sure what exactly it is that's so offensive, i think they're more attractive than linksys |
| 06:08.14 | [TK]D-Fender | dlynes, While the new dislplay IS pixel based, their FIRMWARE is still solidly 1980's :) |
| 06:08.30 | mosty | i'm curious to know what i'm missing out on not having a big lcd on my phone, what kind of "services" do people use them for? |
| 06:08.38 | [TK]D-Fender | mosty, Linksys ones ARE a little harder to read. |
| 06:08.40 | JT | mosty: have you seen the snom 360 lcd? |
| 06:08.44 | [TK]D-Fender | mosty, but the layout is sane. |
| 06:08.50 | JT | it looks like a children's toy |
| 06:09.03 | dlynes | mosty: the user interface on the snom's is horrible |
| 06:09.11 | [hC] | the new snom 370 looks alright |
| 06:09.11 | [TK]D-Fender | mosty, on my work phones I get live Queue stats, weather, etc..... |
| 06:09.14 | dlynes | mosty: it's the most user unfriendly one i've run across |
| 06:09.26 | mosty | JT: i used to have a snom360, or maybe it was a 370, i'm not sure |
| 06:09.34 | l0verb0y | does anyone know how to remove, or change the beep when using the record command? |
| 06:09.46 | [hC] | although the new 5xi series from aastra can do a ton, theres enough about them that drive me mental |
| 06:09.49 | JT | i have a linksys 941 and cisco 7905 here |
| 06:09.50 | dlynes | mosty: but the only ones I can base my experience on is the snom 100, snom 110 and the snom 360 |
| 06:10.01 | JT | both the linksys and cisco have 500% smarter LCD design |
| 06:10.10 | dlynes | [hC]: you mean like the phone just completely locking up for no apparent reason? |
| 06:10.21 | [TK]D-Fender | dlynes, Yup, happened plenty of times with me... |
| 06:10.30 | [hC] | dlynes: ive had that, i am on beta firmware that has fixed it, primarily with BLF. |
| 06:10.40 | *** join/#asterisk waptaxi (i=kvirc@stat-5-160.e-sky.ru) |
| 06:10.41 | dlynes | [hC]: where do you get the beta firmware from? |
| 06:10.53 | [hC] | dlynes: im more annoyed at the physical aspects... the soft shitty buttons, the crappy handset that slips and slides off your shoulder when trying to use it. |
| 06:11.05 | [hC] | dlynes: I am an aastra partner, and deal very closely with their engineers, t hey sent it to me. |
| 06:11.11 | [TK]D-Fender | And the 5i's rubber shit-for-all buttons, char matix'd pixel screen, non-independant DECT, awkrwad button placement |
| 06:11.12 | [hC] | let me check, it may have been released already |
| 06:11.12 | mosty | [TK]D-Fender, ahh well i do that stuff with web services or similar. most people have much better monitors on their computer |
| 06:11.14 | dlynes | [hC]: ah |
| 06:11.32 | [TK]D-Fender | mosty, Mines good "at a glance" so they don't have to switch screens. |
| 06:11.33 | [hC] | nope. |
| 06:11.36 | JT | and the lcd on the snom320 looks about the same as a cisco 12SP+ |
| 06:11.39 | [hC] | im going to have to hassle them |
| 06:11.44 | [TK]D-Fender | mosty, more user efficient. |
| 06:11.49 | dlynes | [hC]: is there a way to get the phone's handset speaker to go any louder? |
| 06:11.49 | JT | which is saying a lot since that phone is a decade old or more |
| 06:11.59 | dlynes | [hC]: I've got plenty of customers complaining it's not loud enough |
| 06:12.10 | [hC] | dlynes: i believe there are gain levels you can set in the aastra.cfg, but i havent had to |
| 06:12.18 | dlynes | [hC]: yeah...those don't do the trick |
| 06:12.27 | dlynes | [hC]: they're more for microphone gains |
| 06:12.32 | [hC] | dlynes: i just turn the volume up to max on the handset. |
| 06:12.40 | [hC] | dlynes: are you using this over zap channels? |
| 06:12.44 | dlynes | [hC]: yeah...that's not loud enough for these deaf mofos |
| 06:12.46 | [hC] | dlynes: maybe you need to look at your zap gains? |
| 06:13.10 | dlynes | [hC]: adjusting the zap gains up, just causes poorer call quality |
| 06:13.26 | [hC] | dlynes: what kind of card? |
| 06:13.27 | [TK]D-Fender | dlynes, buy a better card :) |
| 06:13.35 | dlynes | [hC]: sangoma a400d |
| 06:13.44 | [TK]D-Fender | :/ |
| 06:13.52 | dlynes | that shut [TK]D-Fender up :) |
| 06:14.11 | [TK]D-Fender | dunno... maybe some build flaws.. never used a 400 series |
| 06:14.22 | dlynes | [TK]D-Fender: it's still the same remora board |
| 06:14.23 | [hC] | I use sangoma a200d's in some spots, but try with all my might to avoid ever using analog |
| 06:14.37 | [hC] | it begs to have people bitch and moan about shit and have weird crap happen. |
| 06:14.48 | [TK]D-Fender | dlynes, tested other phones? |
| 06:14.50 | dlynes | [hC]: i don't have a choice...none of our customers have enough lines to warrant a pri |
| 06:15.03 | [TK]D-Fender | dlynes, I found my 57i CT "iffy". |
| 06:15.03 | mosty | dlynes, BRI? |
| 06:15.08 | [hC] | dlynes: and you dont have a clean enough network to deliver them IP trunks? |
| 06:15.10 | [TK]D-Fender | mosty, not in CANADA :p |
| 06:15.18 | mosty | fractional PRI? |
| 06:15.31 | dlynes | [hC]: we do, but even then, the call quality is shitty for ip |
| 06:15.38 | dlynes | [hC]: we're trying to determine why right now |
| 06:15.38 | [hC] | dlynes: whats your company name again? |
| 06:15.41 | JT | mosty: BRI is not an options with asterisk in the USA and Canada. |
| 06:15.46 | JT | s/options/option |
| 06:15.58 | dlynes | [hC]: we're using skyway west for the ip |
| 06:16.00 | JT | NI2 BRI is not supported in Asterisk |
| 06:16.08 | mosty | oh ok |
| 06:16.24 | JT | only ETSI BRI really is |
| 06:16.24 | [hC] | dlynes: heh. funny you should mention that. I use skyway too. and for the last 3 weeks ive had calls go to shit, and i already know why and I'm working with them to get it solved |
| 06:16.54 | [hC] | dlynes: if youve gotten as far as smokepinging or mtr'ing the link you probably see periodic packet loss at the dsl modem. |
| 06:17.08 | [hC] | dlynes: wait.. you dont work for galaxy do you? |
| 06:17.12 | dlynes | [hC]: well, we're working towards getting preferential routing set up iwth them, and to get them to put qos on their end |
| 06:17.15 | dlynes | [hC]: no |
| 06:17.24 | [hC] | dlynes: yeah thats what we're doing with them too :) |
| 06:17.36 | [hC] | dlynes: whats your company again? |
| 06:17.41 | dlynes | [hC]: 24/7 Communications |
| 06:17.54 | [hC] | ah yeah |
| 06:18.06 | dlynes | [hC]: I'm sure you probably know Gurpreet |
| 06:18.42 | [TK]D-Fender | ok, bed time... |
| 06:18.44 | [TK]D-Fender | later all |
| 06:18.57 | [hC] | dlynes: no, but im not surprised, im not a telco guy, im new in this sector here. i came from routing and network security |
| 06:21.28 | dlynes | [hC]: i came from system admin and programming, myself |
| 06:23.05 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-4b98199150bb337f) |
| 06:24.05 | *** join/#asterisk tengulre (n=tengulre@222.90.15.186) |
| 06:25.29 | *** part/#asterisk tengulre (n=tengulre@222.90.15.186) |
| 06:28.08 | obnauticus | How do I record a channel? |
| 06:28.12 | obnauticus | or enable recording |
| 06:29.42 | JT | MixMonitor or Monitor |
| 06:29.53 | mosty | or one-touch recording via features.conf |
| 06:34.18 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 06:34.40 | *** join/#asterisk implicit (n=implicit@ip68-4-84-39.oc.oc.cox.net) |
| 06:35.19 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:35.23 | *** part/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:35.42 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 06:35.45 | GuyOCanada | I somewhere read that you can use asterisk to record your IVR sounds and it would save it |
| 06:36.52 | mosty | Record |
| 06:37.01 | `Sean | mosty one touch recording? |
| 06:37.22 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 06:37.40 | mosty | yes, it's one of the things you can enable in features.conf |
| 06:37.41 | obnauticus | |
| 06:37.46 | `Sean | hrmp |
| 06:37.47 | `Sean | nice |
| 06:38.31 | GuyOCanada | ;automon => *1; One Touch Record a.k.a. Touch Monitor |
| 06:38.38 | GuyOCanada | how does it work? |
| 06:38.51 | mosty | dial *1 during a call to start/stop recording |
| 06:38.53 | `Sean | nice? |
| 06:38.55 | `Sean | err |
| 06:39.09 | `Sean | so during call you just dail *1 while talking to someone and it starts recording? |
| 06:39.16 | `Sean | and how to stop other then hanging up? |
| 06:39.27 | GuyOCanada | its a switch |
| 06:39.28 | `Sean | ahh sorry didn't see your thing screen buffer is being gay |
| 06:39.31 | GuyOCanada | start/stop same again :) |
| 06:39.44 | `Sean | hrmp quite interesting |
| 06:42.35 | GuyOCanada | everytime i pres # x-lite crashes |
| 06:54.18 | GuyOCanada | which is the hash key? |
| 06:55.19 | mosty | # |
| 06:56.18 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 06:57.17 | GuyOCanada | so hash and paund are the same? |
| 06:57.39 | mosty | yes. for some reason american's call it pound, i don't know why |
| 06:58.49 | *** join/#asterisk reber (n=reber@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 06:58.56 | GuyOCanada | :) |
| 06:58.58 | Corydon76-dig | or octothorpe |
| 06:59.09 | GuyOCanada | i have heard people calling it the square key |
| 06:59.33 | GuyOCanada | I got a question lets say i do Background(for-sales&press-1&for-billing&press-2&for&customer-relations&press-3&for&technical-support&press-4) |
| 07:00.04 | GuyOCanada | i have to use for and technical-support as there is no sound that says "for technical support" there is a delay between the for and the technical support |
| 07:00.09 | GuyOCanada | is there a way to remove that delay? |
| 07:00.11 | Corydon76-dig | The reason why it's called the pound key is that Shift-3 is that symbol on American keyboards and the English pound symbol on British keyboards |
| 07:00.49 | mosty | GuyOCanada, you can just record a single sound file |
| 07:01.01 | mosty | or merge them together with a sound program |
| 07:01.47 | mosty | Corydon76-dig, heh they should swap the names for y and z too, since they're swapped on german keyboards |
| 07:02.26 | Corydon76-dig | What, yed and zed? |
| 07:03.35 | GuyOCanada | yes |
| 07:04.04 | GuyOCanada | the german keyboard is a qwerty keyboard but instead of the Z it has Y :) |
| 07:04.34 | Corydon76-dig | anyway, I'm off to bed |
| 07:04.45 | GuyOCanada | can you do decimals in Wait? |
| 07:08.29 | obnauticus | Can someone tell me why this sends me to busy |
| 07:08.29 | obnauticus | exten => 508,1,Dial(sip/*5&sip/*6,15) |
| 07:08.33 | obnauticus | exten => 508,2,Voicemail(911@default) |
| 07:08.37 | obnauticus | exten => 508,3,Hangup() |
| 07:08.56 | obnauticus | It just sends all circuits busy to the caller. |
| 07:09.29 | mosty | what are those *'s in the dial command? |
| 07:09.41 | obnauticus | Those are the extensions it is dialing |
| 07:09.44 | obnauticus | *5 and *6 |
| 07:10.14 | mosty | i would try using sip accounts that don't have *'s in them |
| 07:10.24 | obnauticus | It works internally |
| 07:10.31 | obnauticus | like when i call *5 directly |
| 07:10.33 | obnauticus | if it times out |
| 07:11.38 | mosty | set debug and verbose to 10, then paste the full log on a paste site from when it fails |
| 07:12.01 | obnauticus | k hold |
| 07:12.54 | obnauticus | http://pastebin.ca/783527 |
| 07:13.04 | obnauticus | I ignored the call on purpose |
| 07:13.08 | obnauticus | on the other computer |
| 07:13.24 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 07:13.30 | obnauticus | it doesn't matter wether or not I do, i always get cirecuity busy from *6 when dialing in from extension 508 |
| 07:14.12 | mosty | pastebin.ca just shows me a blank page |
| 07:14.26 | mosty | using firefox, and also with wget :/ |
| 07:14.33 | obnauticus | Nuts. |
| 07:14.35 | obnauticus | that's weird. |
| 07:14.56 | mosty | it was happening to me the other day also, try another paste site |
| 07:14.57 | obnauticus | http://rafb.net/p/ODfFPB99.html |
| 07:15.31 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-7213cf849999bbd0) |
| 07:15.48 | mosty | which of the two extensions is at 10.0.0.110 ? |
| 07:15.54 | obnauticus | *6 |
| 07:16.07 | obnauticus | wait.....ya |
| 07:16.26 | *** join/#asterisk Kumbang (n=kumbang@router-wmi.paume.ITB.ac.id) |
| 07:16.27 | mosty | *6 is in DND mode |
| 07:16.36 | mosty | turn that off |
| 07:16.39 | obnauticus | It rings. |
| 07:16.41 | obnauticus | It's a hardphone. |
| 07:16.44 | obnauticus | Cisco 7960 |
| 07:18.05 | obnauticus | i'll change dnd_conrol: "2" (off with no user control) to just off |
| 07:19.33 | GuyOCanada | anyone using Cepstral voices? |
| 07:21.42 | obnauticus | k |
| 07:21.45 | obnauticus | mosty this is what happens |
| 07:21.49 | obnauticus | right when it says this: |
| 07:21.49 | obnauticus | <PROTECTED> |
| 07:22.08 | obnauticus | It plays busy :| |
| 07:22.30 | mosty | and when you dial *6 by itself? |
| 07:22.57 | obnauticus | I get voicemail after 1500ms of no pickup |
| 07:22.58 | obnauticus | or |
| 07:23.00 | obnauticus | 15000* |
| 07:23.17 | mosty | does it ring? |
| 07:23.20 | obnauticus | Ya. |
| 07:23.31 | mosty | what version of asterisk is this? |
| 07:23.49 | obnauticus | Asterisk SVN-branch-1.4-r71230 built by root @ asterisk on a i686 running Linux on 2007-06-23 00:39:02 UTC |
| 07:23.59 | obnauticus | <PROTECTED> |
| 07:23.59 | obnauticus | <PROTECTED> |
| 07:24.00 | obnauticus | that |
| 07:24.03 | obnauticus | is what it should be doing. |
| 07:24.21 | mosty | i would upgrade to a non-svn build of asterisk |
| 07:24.33 | obnauticus | I was told it was fine :| |
| 07:24.47 | mosty | by who? |
| 07:24.56 | obnauticus | I think it was D-Fender |
| 07:25.02 | obnauticus | in here that wsaid my SVN build was fine some time ago |
| 07:25.08 | obnauticus | [TK]D-Fender |
| 07:25.13 | mosty | i doubt they checked every line of code though |
| 07:25.21 | obnauticus | Who knows |
| 07:25.22 | obnauticus | LOL |
| 07:25.22 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 07:25.41 | mosty | 1.4 has been released, you should probably use the latest 1.4 release |
| 07:25.55 | obnauticus | This is 1.4 |
| 07:25.56 | obnauticus | i think |
| 07:26.02 | obnauticus | just the svn release |
| 07:26.05 | obnauticus | 1.4 r71230 |
| 07:26.25 | mosty | svn revisions aren't releases |
| 07:26.37 | obnauticus | oh :/ |
| 07:26.49 | obnauticus | Could I just donwload the source extract it and recompile really fast? |
| 07:26.54 | mosty | get 1.4.13 or whatever the latest 1.4 release is |
| 07:26.59 | obnauticus | .47 |
| 07:27.01 | obnauticus | 14** |
| 07:31.31 | tzafrir | 1.4.14, you mean |
| 07:31.45 | *** join/#asterisk implicit (n=implicit@ip68-4-84-39.oc.oc.cox.net) |
| 07:32.08 | tzafrir | mosty, check /topic once in a while... |
| 07:33.00 | obnauticus | mosty, what's the most current package version of asterisk-sounds |
| 07:33.06 | obnauticus | and where is the full .tar.gz for it :\ |
| 07:33.16 | mosty | on the digium download site |
| 07:33.31 | obnauticus | It doesn't have a whole asterisk-sounds-xx-xx.tar.gz |
| 07:33.36 | obnauticus | Or I can't find it :| |
| 07:33.56 | obnauticus | ohh |
| 07:33.57 | obnauticus | i think i got it |
| 07:34.02 | obnauticus | asterisk-core-sounds-en-ulaw-current.tar.gz |
| 07:38.07 | *** join/#asterisk syneus (n=syneus@syneus.aemcom.net) |
| 07:48.46 | obnauticus | oo |
| 07:48.51 | obnauticus | mosty festival is included in this version? |
| 07:49.03 | obnauticus | <PROTECTED> |
| 07:49.23 | mosty | i guess so |
| 07:49.50 | *** join/#asterisk agx (n=AGX@host63-216-static.34-88-b.business.telecomitalia.it) |
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| 07:56.42 | tzafrir | obnauticus, no. This app only executes an external festival executable |
| 08:00.02 | obnauticus | Oh |
| 08:00.03 | obnauticus | :| |
| 08:00.07 | obnauticus | mosty, thanks: Asterisk 1.4.14 built by root @ asterisk on a i686 running Linux on 2007-11-18 23:32:29 UTC |
| 08:00.58 | mosty | try it out now, see if you're lucky and the issue no longer occurs |
| 08:01.32 | obnauticus | ya |
| 08:01.34 | obnauticus | still does. |
| 08:01.36 | obnauticus | <PROTECTED> |
| 08:04.00 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:06.18 | tzafrir | obnauticus, please pastebin: dialplan show inbound |
| 08:07.14 | obnauticus | http://rafb.net/p/3jCEPw34.html |
| 08:07.24 | obnauticus | that hangup application on priority 3 was just for debugging purposes. |
| 08:09.14 | tzafrir | obnauticus, I suspect an extra space somewhere |
| 08:09.28 | tzafrir | can you provide the full trace for that call? |
| 08:10.07 | obnauticus | k |
| 08:10.24 | obnauticus | Actually. |
| 08:10.32 | obnauticus | those are the only lines handling the 508 extension |
| 08:10.45 | tzafrir | How exactly do you call it? |
| 08:11.13 | obnauticus | PSTN -> Ipkall -> Intarwebz -> NAT -> * server |
| 08:11.28 | obnauticus | backwards the same way. |
| 08:11.55 | tzafrir | Let's try something simpler: |
| 08:12.21 | *** join/#asterisk mildk (n=mil@duke.code3.dk) |
| 08:12.23 | tzafrir | originate Local/508@inbound application Echo, |
| 08:13.07 | obnauticus | <PROTECTED> |
| 08:13.18 | obnauticus | It's not going to the voicemail like it's supposed to |
| 08:13.19 | obnauticus | :\ |
| 08:13.20 | tzafrir | core set verbose 3 |
| 08:13.23 | tzafrir | and try again |
| 08:13.25 | obnauticus | it's set to the max |
| 08:14.00 | tzafrir | sip show peers |
| 08:14.08 | tzafrir | Do you really have '*5' and '*6'? |
| 08:14.14 | obnauticus | 508 66.54.140.46 5060 Unmonitored |
| 08:14.16 | tzafrir | those are strange names for sip peers |
| 08:14.30 | obnauticus | I'm trying to keep peers on *[num] |
| 08:14.35 | obnauticus | services on ##[num]## |
| 08:15.10 | tzafrir | buy the peer name seems to be 508 (if the above is indeed th output from 'sip show peers') |
| 08:15.57 | obnauticus | ya |
| 08:16.00 | obnauticus | it is. |
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| 08:35.21 | *** join/#asterisk alephcom (n=chatzill@h66-112-187-16.mcsnet.ca) |
| 08:36.13 | alephcom | Ok, I'm baffled. Here's my output from asterisk: |
| 08:36.21 | alephcom | <PROTECTED> |
| 08:36.22 | alephcom | <PROTECTED> |
| 08:36.24 | alephcom | <PROTECTED> |
| 08:36.41 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 08:36.46 | alephcom | any ideas why I only hear the first one? |
| 08:37.32 | alephcom | I'm connected using SJphone and these are played from within an agi script. It's trying to collect data from me but it doesn't seem to get what I enter on some of them. |
| 08:38.45 | alephcom | great, this time I didn't hear any of them. lol |
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| 09:02.33 | mosty | where does "make menuselect" in the asterisk source save it's settings? |
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| 09:26.43 | Somebee | Hi. I have 5 sip-softphones (xlite) connected to an asterisk server (using SIP). Is it best to use SIP or IAX from server to provider? |
| 09:28.31 | agx | i've just released 1.4.2 (fixed a deadlock in app_pickup2.c) http://sourceforge.net/projects/agx-ast-addons/ |
| 09:30.56 | *** join/#asterisk ming_zym (i=ming_zym@nat/yahoo/x-26a8e281d15aea4d) |
| 09:31.37 | JT | Somebee: sip |
| 09:31.45 | mosty | iax uses less bandwidth |
| 09:31.45 | Somebee | ok |
| 09:32.09 | JT | marginally |
| 09:32.19 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:32.21 | JT | unfortunately iax also sucks balls ;) |
| 09:33.06 | Somebee | JT: Why is that? My provider meant that Iax was better, and from searching voip-info I've got the same impression |
| 09:33.06 | mosty | what don't you like about iax? |
| 09:33.51 | JT | it's non standard, asterisk proprietary, poorly implemented in asterisk, does not scale |
| 09:34.12 | mosty | it doesn't need to be "standard" if there's only a single implementation |
| 09:34.26 | JT | there's not a single implementation |
| 09:34.34 | mosty | effectively there is |
| 09:34.34 | JT | anyway, it's about interop, and iax doesn't have it |
| 09:34.37 | JT | and it's unreliable |
| 09:34.40 | JT | nup |
| 09:34.53 | mosty | iax's major downside is pretty much only asterisk supports it |
| 09:34.57 | JT | and most important for ITSPs, does not scale |
| 09:35.05 | JT | and callweaver, and yate, and freeswitch |
| 09:35.25 | mosty | where does it break in terms of scalability? |
| 09:35.35 | JT | the fact that it's implemented poorly |
| 09:35.42 | JT | it chokes after more than a few trunked calls |
| 09:36.00 | JT | and the combination of signalling and media does not scale from a provider viewpoint |
| 09:36.09 | JT | you can't easily stick a proxy in front |
| 09:36.30 | JT | also use of hardware timers for trunking is another bad point |
| 09:36.59 | mosty | i do wish there was something like openser for iax |
| 09:37.31 | JT | the only thing i like about iax is that its call signalling seems closer to Q.931/H.323 than SIP ;) |
| 09:37.49 | mosty | the nat-traversability is handy |
| 09:38.11 | JT | yeah it's not foolproof though |
| 09:38.22 | JT | and sip can easily traverse most NAT devices |
| 09:38.36 | mosty | sip breaks more easily that's for sure |
| 09:38.55 | JT | rarely if set up correctly though ;) |
| 09:39.07 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 09:43.48 | alephcom | That's interesting. When I turn all debugging off on my agi script it suddenly works. |
| 09:46.57 | agx | alejandro, perhaps your sending something to STDIN by mistake :) |
| 09:48.13 | *** join/#asterisk santoshr (i=1063@203.199.110.93) |
| 09:49.37 | santoshr | I have an ATA which can handle three sip accounts. Two different sip accounts on an asterisk service register from the same IP. But when i try to call. It says check_auth: username mismatch, have <24>, digest has <22> |
| 09:50.57 | mosty | sounds like a bug in the ata |
| 09:52.35 | santoshr | No but if i send the second SIp a/c to a different asterisk box. the call goes through. |
| 09:53.04 | santoshr | i feel asterisk is getting confused since both a/c's have come from the id. |
| 09:53.07 | santoshr | ip sorry |
| 09:53.10 | mosty | yes, it sounds like the ata has a bug with two accounts on the same server |
| 09:53.33 | JT | what is the ata? |
| 09:53.51 | mosty | hmm, can you choose local sip ports on the ata? maybe set different ports for each account |
| 09:54.20 | JT | that's not an ata bug i think |
| 09:54.40 | JT | asterisk's chan_sip cracks the shits with multiple registrations to the one ip |
| 09:54.44 | JT | last i checked |
| 09:54.53 | JT | it can't correctly identify them |
| 09:55.13 | mosty | i think it can if you choose different local ports for each sip client |
| 09:55.17 | santoshr | different local sip ports not an option |
| 09:55.26 | JT | again |
| 09:55.29 | JT | what ATA is it? |
| 09:55.46 | santoshr | its a welltech device |
| 09:55.49 | santoshr | ata171 |
| 09:56.06 | JT | ah dodgy :P |
| 09:56.59 | santoshr | meaning |
| 09:57.09 | JT | not good, cheap |
| 09:58.00 | santoshr | Is this a device bug or asterisk |
| 09:58.10 | JT | not sure |
| 09:58.23 | oej | As long as each registration has a different user name (account) we should be fine |
| 09:58.57 | santoshr | i checked the debug the invite was a/c specific. |
| 09:59.16 | oej | So what kind of object do you have in sip.conf for them? |
| 09:59.23 | santoshr | realtime |
| 09:59.37 | oej | yes, but what kind of object? peer, user, friend |
| 09:59.46 | mosty | is the ata behind nat? |
| 10:00.28 | santoshr | no .. ata not behind nat |
| 10:00.40 | santoshr | it has a public ip. |
| 10:01.00 | santoshr | oej: wht object is ti by default. because we have not mentioned it in the db |
| 10:01.40 | oej | Then it's a peer, and calls from peers are matched on IP. So you need a user for each device. Check the docs for realtime users |
| 10:02.02 | oej | If you add a user for each account on the device, Asterisk will match on the username and separate them. |
| 10:02.16 | oej | Realtime is not a good starting point for learning basic Asterisk stuff. It's a kludge. |
| 10:08.59 | *** join/#asterisk af_ (n=getsmart@88-149-240-124.dynamic.ngi.it) |
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| 10:55.13 | l0verb0y | hey |
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| 11:32.40 | zeeesh | still i could not attached vicemail in email .. how to do it ? |
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| 11:52.24 | J4zen | Any dutch people know a good place for SNOM320's or other VOIP material(quality/price)? |
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| 11:52.36 | masus | hi all, load balance 2 inet connections and use it for sip is someting like this possible ? Thanks |
| 11:54.32 | harpal | I have just one PC installed Linux and I want to set up PBX What should I need. I have installed asterisk on linux. |
| 11:55.27 | masus | harpal: you need to set up asterisk |
| 11:55.40 | harpal | Do I need Zaptel Drivers? I dont have that cards. Is it necessary to install card |
| 11:56.26 | harpal | masus, ok. I have installed asterisk. now I have to configure it right? |
| 11:56.35 | masus | yep |
| 11:57.06 | J4zen | Does anyone have any expierence regarding the SNOM320 vs. SNOM360? is the 360 an actual upgrade or? |
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| 11:57.45 | harpal | masus, So I can connect it with softphone and use it? or I need some thing? |
| 11:57.58 | masus | harpal: thats enough ... |
| 11:58.02 | Arkonek | hello, if im getting timing source auto card 0! |
| 11:58.07 | masus | u can try it without cards |
| 11:58.29 | masus | for example u can register 2 softphones, and speak with each other |
| 11:58.35 | Arkonek | hello, if im getting timing source auto card 0 does it mean that timing comes form my teleco or is generetad by my card? |
| 11:59.25 | masus | harpal: u have to config these files , sip.conf , extensions.conf |
| 11:59.26 | tzafrir | Arkonek, what card? |
| 11:59.40 | Arkonek | TE210p after runing zaptel |
| 12:00.21 | bobkare | harpal: I think you need ztdummy for some timing or something |
| 12:01.08 | harpal | masus, ok. |
| 12:01.30 | harpal | bobkare, Is it a packet to install? |
| 12:01.46 | bobkare | what distro? |
| 12:02.06 | harpal | I have debian |
| 12:03.52 | Arkonek | so? what with this timing?:) |
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| 12:05.10 | harpal | masus, Do I need Asterisk-sound? |
| 12:06.08 | bobkare | harpal: yes, zaptel-source, then module-assistant a-i zaptel |
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| 12:08.29 | DrukenLPY | morning everyone |
| 12:08.51 | hi365 | im having a problem streaming MOH using the folowing comand: application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://209.9.229.207:8080 |
| 12:09.20 | hi365 | is it writen wrongly? |
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| 12:10.28 | DrukenLPY | what's it not doing? |
| 12:13.59 | bobkare | Can I use AMI to add phones to chan_mobile's mobile.conf? |
| 12:15.02 | masus | harpal: for test use not |
| 12:15.17 | masus | harpal: it's already installed with asterisk |
| 12:15.25 | Arkonek | help me please, when loading ztcfg and then watching /var/log/messages i can see timing source auto card 0! does it mean that something is wrong with synchro with my teleco company? |
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| 12:16.50 | harpal | masus, thanks |
| 12:18.07 | masus | harpal: cd /var/lib/asterisk/sounds/ |
| 12:20.11 | harpal | masus, than? |
| 12:20.21 | tzafrir | Arkonek, I think it is OK |
| 12:22.21 | masus | harpal: no i mean it's there |
| 12:22.24 | masus | asterisk sounds |
| 12:23.45 | harpal | masus, ok I have that directory and some files there |
| 12:24.06 | Arkonek | tzafrir, so it mean that card is working well with my operator? |
| 12:24.18 | santoshr | oej: is it allowed for the type to come from the Db in relatime |
| 12:24.24 | santoshr | *realtime |
| 12:24.48 | harpal | masus, I have installed asterisk so now I need to configure asterisk to test that, right? |
| 12:24.55 | oej | no, but there's two different tables for peers and users, santoshr |
| 12:25.18 | tzafrir | Arkonek, I don't think that this is a sign of a problem |
| 12:26.00 | santoshr | sippeers and sipusers |
| 12:26.03 | Arkonek | ok, thx |
| 12:26.39 | santoshr | if i give sipusers in the extconfig.conf. it doesnt registr the device. I see the query happening on the DB but somehow it doent do anything with it |
| 12:27.14 | santoshr | querying sippeers and type being in the result set as "user" ... is this allowed |
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| 12:33.05 | hi365 | im having a problem streaming MOH using the folowing comand: application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://209.9.229.207:8080. Its not playing. in the cli i get: started music on hold and imideatly stop music on hold |
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| 12:39.04 | J4zen | welcome back |
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| 12:40.41 | yang | J4zen: yes i noticed, that is why i am repeating the third time |
| 12:40.48 | yang | I am wondering if there is some GUI interface management for asterisk like HUD pro which works on trixbox? |
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| 12:42.28 | J4zen | Asterisk on voip.isaeus.nl exited on signal 11. Might want to take a peek. |
| 12:42.28 | yang | they keep dropping in |
| 12:42.28 | J4zen | yeah |
| 12:42.28 | J4zen | outdated clients probably |
| 12:43.14 | hi365 | im having a problem streaming MOH using the folowing comand: application=/usr/local/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0 http://209.9.229.207:8080. Its not playing. in the cli i get: started music on hold and imideatly stop music on hold |
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| 12:44.57 | santoshr | it sippeers in extconfig.conf is made to sipusers. it doesnt query the table at all |
| 12:45.02 | santoshr | i want some ata's to be peer and the others as users how can tht be done |
| 12:45.10 | santoshr | querying sippeers and type being in the result set as "user" ... is this allowed |
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| 12:46.44 | santoshr | after changing the extconfig.conf .. do we have to unload and load the dsn again ?? |
| 12:47.04 | *** join/#asterisk grantm (n=grant@kolob.wingateservices.com) |
| 12:47.11 | *** join/#asterisk KermitTheFragger (n=siepkes@118-197.bbned.dsl.internl.net) |
| 12:49.20 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 12:50.00 | *** join/#asterisk jetlagmk2 (i=jetlag@pool-70-106-89-254.hag.east.verizon.net) |
| 12:51.50 | *** join/#asterisk deever (n=deever@217-20-127-116.internetserviceteam.com) |
| 12:51.58 | deever | hi |
| 12:52.28 | *** join/#asterisk BeeBuu (n=chatzill@219.130.252.147) |
| 12:53.07 | BeeBuu | please,any one teach me where can i download asterisk-addon? |
| 12:53.42 | deever | BeeBuu: from the asterisk site? ;) |
| 12:53.56 | BeeBuu | any where you like |
| 12:54.09 | BeeBuu | i know no one... |
| 12:55.13 | BeeBuu | deever: have you get one? |
| 12:56.52 | deever | when i try to install asterisk into my homedir (./configure --prefix=/home/deever'), target 'datafiles' stops with error 1: |
| 12:56.55 | deever | mkdir -p /var/lib/asterisk/static-http |
| 12:56.55 | deever | mkdir: cannot create directory `/var/lib/asterisk': Permission denied |
| 12:57.20 | deever | BeeBuu: http://www.asterisk.org ? |
| 12:57.34 | BeeBuu | http://downloads.digium.com/pub/asterisk/ |
| 12:57.58 | rob0 | You just have to wonder about people who come in here and ask things like "where can i download ..." Like ... where did you LOOK? |
| 12:58.46 | *** join/#asterisk chesty (n=chesty@unconcerned.org) |
| 12:59.02 | rob0 | deever: asterisk.conf has paths. Change those to suit. |
| 12:59.29 | deever | rob0: thx, i'll try it out! :) |
| 12:59.53 | rob0 | deever: There's also a page on the wiki about running as nonroot. |
| 13:00.29 | rob0 | Most of that probably applies to Zaptel device permissions, which might not be an issue if you are zaptelfree. |
| 13:01.05 | rob0 | oh!! You had compile-time troubles. |
| 13:01.46 | rob0 | No, I don't know how to fix that, except maybe with more ./configure --options. |
| 13:03.39 | hi365 | can anyone help with streaming moh? |
| 13:04.14 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 13:05.04 | *** join/#asterisk Telemac (n=telemac@213.223.113.74) |
| 13:05.10 | Telemac | Hello |
| 13:05.29 | *** join/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 13:05.56 | santoshr | querying realtime sippeers and type being in the result set as "user" ... is this allowed |
| 13:06.38 | santoshr | oej: u around bro |
| 13:07.02 | *** join/#asterisk Somebee (n=Somebee@80.232.5.97) |
| 13:08.04 | coppice | there are huge numbers of Z80s being made. It wouldn't surprise me to find some running CP/M |
| 13:09.06 | Telemac | I've read oreilly book and asterisk guru tutorial about dynamic realtime. It works perfectly for sip and iax users but not for voicemail. I think there is trouble about table definition in PostgreSQL, could anyone who has successed in that points me to the proper table def ? |
| 13:09.34 | cpm | it's been on loan for a decade or so, but as far as I know, it's still around, and still works. |
| 13:10.07 | cpm | http://www.mcsquared.com/tef.htm |
| 13:10.44 | *** join/#asterisk shido6 (n=shido6@204.126.120.132) |
| 13:13.16 | *** join/#asterisk Greek-Boy (n=email@41.221.58.2) |
| 13:15.58 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 13:16.25 | coppice | you can get a loan, where the lender forgets about it for a decade? can you get the same deal with a mortgage? |
| 13:18.15 | [TK]D-Fender | coppice: not by definition (or better yet, direct translation) of the word "mortgage" |
| 13:20.07 | coppice | in these days of credit crunch you gotta take what you can get |
| 13:20.08 | rob0 | I actually did run across an apparently-forgotten mortgage. They tried to foreclose too late. |
| 13:20.25 | rob0 | ($DAYJOB is land title research.) |
| 13:20.57 | coppice | i think its only too late to foreclose when the land sinks beneath the waves |
| 13:21.18 | rob0 | ha, no waves here. |
| 13:21.52 | [TK]D-Fender | rob0: We'll see about that... |
| 13:22.37 | rob0 | Raise the Atlantic a few hundred feet, and I'm in a swimming pool. |
| 13:23.20 | coppice | I'll stick with the swimming pool I can see on the seafront, 50m below me |
| 13:24.11 | cpm | rob0, you do land abstracting work? where? |
| 13:24.56 | coppice | we bought on a hill overlooking the sea, because we all know we can trust Al Gore. the man's a politician, after all |
| 13:25.12 | J4zen | lol |
| 13:25.17 | hi365 | how long after a moh stream is the stream disconected? |
| 13:25.31 | rob0 | Arkansas, and not exactly abstracting, but similar. |
| 13:25.52 | *** join/#asterisk Zuchmir (n=dddddd@123-2-65-142.static.dsl.dodo.com.au) |
| 13:26.03 | harpal | I have installed asterisk and done make samples and make config. now When I try to start asterisk using /etc/init.d/asterisk start than nothing happen. asterisk not seems to run. Whats problem? |
| 13:26.12 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 13:27.30 | [TK]D-Fender | harpal: try via "asterisk -gvvvc" and if it fails, pastebin the complete output of your attempt |
| 13:27.32 | [TK]D-Fender | ~pb |
| 13:27.33 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:27.34 | [TK]D-Fender | ^^^^^^^^^^^^^^ |
| 13:27.39 | cpm | Arkansas eh? figures |
| 13:27.50 | harpal | ok |
| 13:27.54 | *** join/#asterisk ESCulapio__ (n=Esculapi@66.44.88.200.l.sta.codetel.net.do) |
| 13:28.51 | harpal | It shows me asterisk ready. and *CLI> prompt |
| 13:29.45 | santoshr | querying realtime sippeers and type being in the result set as "user" ... is this allowed |
| 13:30.34 | hi365 | anyone know how long after a moh stream is the stream disconected? |
| 13:30.44 | [TK]D-Fender | harpal: and when you do "core show version"? |
| 13:30.50 | *** join/#asterisk grEvenX (n=even@1ult2p8.ip.hipercom.no) |
| 13:31.03 | santoshr | [TK]D-Fender: can i have some set of sip a/c as users and others as peers. Using realtime asteirsk |
| 13:31.15 | *** join/#asterisk DarkDlx (n=darkdll@171.pool85-53-216.dynamic.orange.es) |
| 13:31.34 | [TK]D-Fender | santoshr: No idea, and please don't jsut target people with your questiosn repeating what we saw mere minutes ago |
| 13:31.37 | harpal | [TK]D-Fender, Asterisk 1.4.13 built by root @ debian on a i686 running Linux on 2007-11-07 15:48:42 UTC |
| 13:31.52 | [TK]D-Fender | harpal: then * is fine and you just need a better init script. |
| 13:32.05 | [TK]D-Fender | harpal: I'm not sure what that takes for Debian |
| 13:32.26 | santoshr | point duly noted. I just thought you would remember me. we have spoken alot of times. My bad apologies |
| 13:33.14 | harpal | When i do make config it has added startup script. That script are in config/init.d directory for each distro |
| 13:33.32 | santoshr | and yes the second question was different from the first one. |
| 13:33.59 | *** join/#asterisk ming_zym (n=ming_zym@124.14.234.233) |
| 13:34.30 | harpal | how to quit from that CLI> prompt? quit and exit not working |
| 13:34.35 | *** join/#asterisk stephbul (n=stephbul@ks33394.kimsufi.com) |
| 13:34.47 | DarkDlx | hi sorry for my bad english, im trying to set new audio files for asterisk, i'm connecting through ssh over lan, can i copy my gsm files from an another pc to asterisk pc? or i need to copy it to an usb pendrive, mount it and copy files? |
| 13:35.15 | bobkare | Has anybody used Asterisk::Manager? The way I read the AMI spec and the source the module has to be horribly broken |
| 13:35.19 | [TK]D-Fender | harpal: "stop now" |
| 13:35.26 | *** join/#asterisk rantsh (n=rantsh@190.36.185.139) |
| 13:35.38 | rantsh | Hi all |
| 13:35.51 | [TK]D-Fender | harpal: I know that "make config" does RH scripts, not sure about Debian. Go Google it, I'm sure you'll find the answer very fast. |
| 13:36.28 | harpal | [TK]D-Fender, ok. I am checking that on google |
| 13:36.37 | [TK]D-Fender | DarkDlx: pick whatever method you want so long as they end up in the right place with the right authority, and the right format. |
| 13:37.03 | rantsh | I'd like to setup my asterisk box as a gateway, is there any way I can measure the QoS of a certain route or channel? |
| 13:37.32 | DarkDlx | D-Fender thanks, i ask google |
| 13:39.03 | [TK]D-Fender | rantsh: Forget about QoS over the public internet. |
| 13:39.19 | *** join/#asterisk emk (n=emk@212.49.87.126) |
| 13:39.43 | rantsh | [TK]D-Fender, mmm.... that bad, isn't it? |
| 13:40.39 | harpal | [TK]D-Fender, Its now running. actually I have done /etc/init.d/asterisk restart and restart doesnt do stopping and starting it again. it just reload that things. so Its not starting in restart. I have done /etc/init.d/asterisk start and it starts |
| 13:40.58 | rantsh | [TK]D-Fender, then is there a way I can setup a gateway software such as Quintum's where I can measure which route to send to which provider based on the QoS ? ? ? |
| 13:41.14 | [TK]D-Fender | rantsh: no idea. |
| 13:41.48 | harpal | [TK]D-Fender, thanks a lot |
| 13:41.52 | lirakis | rantsh: ... its the public internet... if you save 2ms on your LAN b/c you implement qos thats fine.. but that same 2ms means very little on the public network.. b/c you have no control over where your packet travels. |
| 13:42.03 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 13:42.06 | rantsh | [TK]D-Fender, :-p dead end!, thanks anyway though |
| 13:42.55 | J4zen | Hm |
| 13:43.05 | J4zen | my outbound SIP-Trunk all the suddon states "All circuits are busy now" |
| 13:43.07 | emk | hi all. Is it an absolute must to buy a Digium card to setup an asterisk server (interfacing between a PSTN panasonic PBX and some Voip Phones) |
| 13:43.09 | J4zen | 5 minutes ago it worked fine |
| 13:43.11 | J4zen | no active lines |
| 13:43.15 | J4zen | or connections |
| 13:43.23 | J4zen | restart gracefully didn't do it |
| 13:43.34 | J4zen | Does anyone have a clue what could caus this behaviour? |
| 13:44.26 | santoshr | the realtime wiki says I can have a type column which will give the type "user" "peer" or "friend". But it does not say whether type=user is allowed from sippeers |
| 13:49.31 | *** join/#asterisk jtexter3 (n=jamest@nat.bloommg.com) |
| 13:57.35 | [TK]D-Fender | emk: If you want to interface you must have some sort of special hardware to allow * access to such things. Not necessarily Digium's. |
| 13:57.56 | *** join/#asterisk JayTee52 (n=jforde05@207-67-84-181.static.twtelecom.net) |
| 13:58.03 | [TK]D-Fender | J4zen: You've shown us nothing of value. Pastebin a failed call with SIP debug enabled |
| 13:58.05 | [TK]D-Fender | ~pb |
| 13:58.06 | jbot | A Pastebin is a place to paste your stuff without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 13:58.07 | [TK]D-Fender | ^^^^^^^^^^^^^^^ |
| 13:58.45 | Greek-Boy | where is the best place in the dial plan to place macros? at the top? |
| 13:59.46 | [TK]D-Fender | Greek-Boy: Doesn't matter. For the sake of creating some kind of structure I try to put mine above anything that will call them. |
| 14:00.16 | JayTee52 | same here |
| 14:01.51 | *** join/#asterisk orsonork (n=orsonork@190.128.168.24) |
| 14:01.55 | orsonork | hello |
| 14:02.22 | orsonork | i have an extension |
| 14:02.28 | orsonork | that dials a number |
| 14:03.12 | orsonork | exten => 555,1,Dial(IAX2/account@something/02312323) |
| 14:03.21 | orsonork | how can i play a sound on that call? |
| 14:03.24 | orsonork | i tried |
| 14:03.28 | orsonork | using |
| 14:03.39 | orsonork | exten => 555,1,Playback(sound) |
| 14:03.40 | orsonork | then |
| 14:03.49 | [TK]D-Fender | orsonork: What exactly are you looking to do? |
| 14:03.57 | orsonork | play a sound |
| 14:03.59 | orsonork | on a call |
| 14:04.05 | destructure | while the call is bridged? |
| 14:04.09 | orsonork | yes |
| 14:04.13 | [TK]D-Fender | orsonork: automatically? are you actually lookijng to TALK to them? |
| 14:04.20 | orsonork | automatically |
| 14:04.23 | orsonork | i already ca TALK to them |
| 14:04.27 | destructure | how far into the call? |
| 14:04.34 | [TK]D-Fender | orsonork: ^^^^ |
| 14:04.43 | orsonork | i can already talk "them" |
| 14:04.46 | orsonork | it's working |
| 14:04.49 | orsonork | i tried using |
| 14:04.51 | destructure | or rather, how do you decide to play the message? |
| 14:04.55 | orsonork | exten => 555,1,Playback/sound) |
| 14:04.56 | orsonork | automatically |
| 14:04.58 | [TK]D-Fender | orsonork: Answer his question |
| 14:05.11 | orsonork | oops |
| 14:05.13 | [TK]D-Fender | orsonork: So you want it to play first, THEN continue talking normally? |
| 14:05.22 | orsonork | yes |
| 14:05.26 | orsonork | i already do that |
| 14:05.28 | orsonork | but it plays |
| 14:05.31 | orsonork | *it plays |
| 14:05.33 | orsonork | for me |
| 14:05.56 | orsonork | but the other call doesn't hear the sound |
| 14:05.56 | destructure | how about taking a step back and describing the call scenario in non-asterisk terms |
| 14:05.59 | [TK]D-Fender | orsonork: thats because you can't use Playback like that. it has to be PART of the dial istself. So go read its instructions again. |
| 14:06.08 | orsonork | Background? |
| 14:06.11 | destructure | nope |
| 14:06.11 | [TK]D-Fender | orsonork: "core show application dial" |
| 14:06.24 | *** join/#asterisk Grizzy (n=Generic@adsl-76-204-25-123.dsl.pltn13.sbcglobal.net) |
| 14:06.25 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:06.30 | destructure | you are misunderstanding the concept of applications executing on a channel |
| 14:06.33 | Greek-Boy | [TK]D-Fender: thanks. your structure makes sense. |
| 14:06.40 | destructure | background is one application. dial is another |
| 14:07.13 | *** join/#asterisk captiancrash (n=jonmoore@70.159.118.70) |
| 14:07.15 | [TK]D-Fender | Greek-Boy: its only purpose is so I know ehere to look. If you wanted you could split up your dialplan into multiple files depending on if you can find sensible bits to break off. |
| 14:07.46 | [TK]D-Fender | Greek-Boy: Just that whatever methos you use actually adds some productivity back, because it all just executes the same in the end |
| 14:08.04 | *** join/#asterisk JackEStorm (n=no@ip68-225-77-136.no.no.cox.net) |
| 14:08.57 | destructure | orsonork: the dialplan executes one app at a time, so if you have bridge a call with dial, no other apps can execute on that channel. However, many tricks exist to work around this, but which you choose depends on your requirements |
| 14:09.48 | J4zen | [TK]D-Fender: http://pastebin.com/m76b4ef84 |
| 14:10.28 | J4zen | A log on verbosity 20 |
| 14:10.28 | J4zen | when trying to use the outbound trunk |
| 14:10.56 | [TK]D-Fender | J4zen: I said *SIP DEBUG ENABLED* |
| 14:11.05 | J4zen | Ah |
| 14:11.10 | J4zen | Pardon, one moment please. |
| 14:13.04 | Greek-Boy | got it TK :) |
| 14:13.15 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584064.dsl.bell.ca) |
| 14:13.45 | J4zen | [TK]D-Fender: http://pastebin.com/m5dfd127e |
| 14:13.47 | J4zen | Updated |
| 14:14.01 | JackEStorm | "Unable to handle return result on switchtype 1!" ...thats bad right? |
| 14:14.54 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 14:14.54 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 14:15.45 | [TK]D-Fender | J4zen: Bad debug -- Executing Dial("SIP/104-0920a468", "SIP/gntel/0641735171|300|") in new stack <- I see no output for this call attempt |
| 14:16.11 | [TK]D-Fender | J4zen: Do not attempt to merely debug your PHONE |
| 14:16.14 | *** join/#asterisk Darthclue (n=chatzill@zeus.nisd.net) |
| 14:16.26 | jtexter3 | I have had an issue 2 times now, wondering if anyone has any thoughts. It's asterisk 1.4.13, Zaptel 1.4.6, two TE412 cards in. Somewhere along the way, when a meetme room is created, it logs failed to open '/dev/zap/pseudo' : No such device or address. A restart of Asterisk solved the issue for about 24 hours, and then it happened again. The only thing that has to be done to resolve it is to restart Asterisk |
| 14:16.37 | jtexter3 | Anyone seen anything like that before? |
| 14:19.00 | *** join/#asterisk _x86_ (n=x86@i.am.leet.org) |
| 14:19.39 | [TK]D-Fender | jtexter3: Make sure to remove ztdummy from your kernel modules "rmmod ztdummy", and redo "ztcfg -vvvv" then restart *. |
| 14:20.12 | [TK]D-Fender | jtexter3: I've head cases where it can interfere. You shouldn't need it because you have a hardware timing source. Also ensue that each card is getting its own IRQQ |
| 14:20.17 | jtexter3 | [TK]D-Fender: I have confirmed ztdummy is not running by doing lsmod | grep ztdummy. Any other thoughts? |
| 14:20.50 | tzafrir | jtexter3, you use centos4, right? |
| 14:20.50 | jtexter3 | [TK]D-Fender: Ah, good point, I shall check IRQ's. I did run zttest last night, and it was hitting at 99.99 |
| 14:21.01 | jtexter3 | tzafrir: centos5 |
| 14:21.31 | J4zen | [TK]D-Fender: http://pastebin.com/m3df9e460 |
| 14:21.34 | J4zen | My final log :) |
| 14:21.43 | J4zen | The last line shows that the trunk returns congestio |
| 14:21.44 | J4zen | n |
| 14:21.55 | tzafrir | Because the handler for /dev/zap/pseudo is the module zaptel itself |
| 14:22.01 | J4zen | eventhough its hard to believe, there are no active channels |
| 14:22.59 | [TK]D-Fender | J4zen: I want to see the ENTIRE call. Stop providing little bits & pieces. |
| 14:23.13 | Greek-Boy | [TK]D-Fender: I need your help with something pls. Can I paste three-liners here? |
| 14:23.20 | [TK]D-Fender | Greek-Boy: Sure |
| 14:23.39 | Greek-Boy | i have a macro like so |
| 14:23.41 | Greek-Boy | [macro-monitor] |
| 14:23.41 | Greek-Boy | exten => s,1,Set(INTERNAL_REC_CALLFILENAME=internal-${CALLERIDNUM}to${EXTEN}-${TIMESTAMP}) |
| 14:23.41 | Greek-Boy | exten => s,2,Monitor(wav,${INTERNAL_REC_CALLFILENAME},m) |
| 14:23.41 | Greek-Boy | exten => s,3,Dial(${ARG1},,r) |
| 14:23.48 | *** join/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 14:23.48 | Greek-Boy | oops, thats 4 lines. sorry |
| 14:24.01 | J4zen | [TK]D-Fender; That is all my debug-window can show. Is there any way to make "sip debug" output all the data to a log? |
| 14:24.15 | [TK]D-Fender | J4zen: SSH in with a better client. |
| 14:24.18 | Greek-Boy | and I call the macro like so: |
| 14:24.18 | Greek-Boy | exten => 100,1,Macro(monitor,SIP/lg-reception01&SIP/lg-reception02&SIP/lg-reception03) |
| 14:24.21 | J4zen | Im using Putty |
| 14:24.26 | J4zen | what wuold you define as a better client? |
| 14:24.32 | [TK]D-Fender | Greek-Boy: Stop using 1.2 deprecated CID vars. |
| 14:24.36 | jtexter3 | tzafrir: My only thought is maybe I had too many file handles open on that particular device, but don't have any evidence. It last happened on Saturday, so saturday night I stopped asterisk, unloaded all zaptel modules, and then brought the system up clean |
| 14:24.43 | J4zen | ? ssh |
| 14:24.46 | [TK]D-Fender | Greek-Boy: and "r" = EVIL. |
| 14:24.56 | Greek-Boy | hmm |
| 14:24.59 | tzafrir | telnet |
| 14:24.59 | Nugget | telnet is eeeeeeevil! |
| 14:25.00 | Greek-Boy | I think i found my problem |
| 14:25.09 | Greek-Boy | let me read UPGRADE.TXT once again |
| 14:25.10 | Greek-Boy | lol |
| 14:25.11 | Greek-Boy | sorry TK |
| 14:25.14 | [TK]D-Fender | J4zen: Screw your trixbox HTTP served up terminal window and connect to your box with a PROPER SSH client. |
| 14:25.36 | [TK]D-Fender | J4zen: If you are incapable of doing this then go learn Linux. |
| 14:26.43 | J4zen | [TK]D-Fender: In stead of making assumptions, read what i said; Im connecting through PUTTY. nowhere did i mention a HTTP served up terminal window as u described |
| 14:26.45 | Greek-Boy | [TK]D-Fender what would you recommend I substitute r with then? for internal context |
| 14:27.14 | *** join/#asterisk [pluto123] (n=cicici@62.123.145.91) |
| 14:27.24 | [TK]D-Fender | J4zen: If you're using PuTTY, then use your SCROLL back. |
| 14:27.28 | [pluto123] | hello to all |
| 14:27.34 | [TK]D-Fender | J4zen: And if you don't have enough, make it BIGGER |
| 14:28.18 | [TK]D-Fender | J4zen: And your'e right it was an assumption. My bad. Then again... so is even asking about problems in here while using a GUI at all... |
| 14:28.19 | tzafrir | And if you don't have a proper terminal and still need a way to scroll back, install screen and use it |
| 14:28.35 | [TK]D-Fender | tzafrir : PuTTY has a very nice scroll-back |
| 14:28.38 | J4zen | [TK]D-Fender: You're right, it is. My bad ;) |
| 14:28.39 | Greek-Boy | UPGRADE.txt says that I should use the dialplan functions instead of variables. Now I gotta find which ones specifically |
| 14:28.49 | tzafrir | IIRC I used screen once or twice with ajaxterm... |
| 14:29.02 | J4zen | The scrolback isn't sufficient for the output it provides |
| 14:29.05 | J4zen | but ill log it |
| 14:29.06 | [TK]D-Fender | J4zen: Good, so this is our last try. Fix your scroll-back capture EVERYTHING and we'll see. |
| 14:29.08 | J4zen | and upload that |
| 14:29.29 | [TK]D-Fender | J4zen: ENLARGE your scrollback. I have mine set to 2000 lines. |
| 14:30.22 | J4zen | Ah, i overlooked that functionality. Thanks |
| 14:30.28 | DarkDlx | where is the ftp folder? |
| 14:30.42 | DarkDlx | not inside srv/ftp/pub/ |
| 14:31.51 | [TK]D-Fender | DarkDlx: What "ftp folder"? The one with the files that CONFIGURE an FTP server on your OS? What OS? What FTP server? HUH!?!? |
| 14:32.01 | DarkDlx | sorry |
| 14:32.05 | [TK]D-Fender | ~cluebat DarkDlx |
| 14:32.06 | jbot | ACTION pulls out a ClueBat (tm) and thwaps DarkDlx. |
| 14:32.06 | DarkDlx | asterisknow |
| 14:32.15 | DarkDlx | the default ftp |
| 14:32.26 | [TK]D-Fender | DarkDlx: Please read the channel topic. That is not supported in this channel. |
| 14:32.34 | DarkDlx | ok sorry again |
| 14:34.11 | *** part/#asterisk santoshr (i=1063@203.199.110.93) |
| 14:34.33 | *** join/#asterisk blitzrage (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 14:34.33 | *** mode/#asterisk [+o blitzrage] by ChanServ |
| 14:34.39 | *** join/#asterisk riddlebox (n=james@75-128-170-26.static.stls.mo.charter.com) |
| 14:34.54 | *** join/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 14:35.24 | Greek-Boy | in 1.4 is the a record application that records both channels in mp3? |
| 14:35.43 | deeperror | mixmonitor + sox? |
| 14:35.56 | J4zen | [TK]D-Fender: Last but not least: http://pastebin.com/m717e1bb0 |
| 14:36.26 | [TK]D-Fender | Greek-Boy: You can't. * cannot ENCODE in MP3 |
| 14:36.57 | Greek-Boy | hmmm |
| 14:37.19 | Greek-Boy | I kinda like the script on the wiki /usr/local/bin/2wav2mp3 |
| 14:37.20 | deeperror | any clues why I would hear a beep come over a zaptel channel when another line is hanging up...the beep is louder when the channel is hearing DT but can still be heard when in a call? |
| 14:37.38 | Greek-Boy | but it has to be modified to work with 1.4 |
| 14:37.42 | [TK]D-Fender | J4zen: SIP/2.0 407 Proxy Authentication Required <-- looks like your peer isn't set up right and you failed the proxy auth. Go ask them how it should be set up in TB |
| 14:37.58 | Greek-Boy | I like it because it puts each channel on its own side of the stereo recording |
| 14:38.35 | [TK]D-Fender | Greek-Boy: need a better mix option. |
| 14:39.16 | [TK]D-Fender | Greek-Boy: MP3 sucks anyways. Its meant to compress alrge WAV's in stereo with an "averaged" center channel. Inappropriate for telephony recordings. |
| 14:39.29 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 14:39.54 | tzafrir | stkn|work, here? |
| 14:41.24 | Greek-Boy | [TK]D-Fender what do u recommend? GSM? |
| 14:42.04 | [TK]D-Fender | Greek-Boy: I like "wav" personally. it is still pretty small (this is not CD quality wav you know...) and is commonly readable |
| 14:42.16 | [TK]D-Fender | Greek-Boy: MP3 adds complications. |
| 14:43.45 | Greek-Boy | [TK]D-Fender: so you only use mixmonitor and leave the recordings in its form as generated by * |
| 14:44.26 | coppice | [TK]D-Fender: of course. you can't do good compression without complex arithmetic :-) |
| 14:45.15 | [TK]D-Fender | coppice: And I've only got so many fingers and toes! |
| 14:45.33 | *** join/#asterisk ronr (n=ron@ip51cdd509.speed.planet.nl) |
| 14:45.43 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:45.56 | coppice | that's OK. the other bits are imaginary anyway |
| 14:46.40 | orsonork | i didn't found any way |
| 14:46.43 | orsonork | to do that |
| 14:46.54 | *** join/#asterisk eserra (i=nobody@89-96-52-24.ip10.fastwebnet.it) |
| 14:47.12 | ronr | we're moving to an asterisk PBX, currently we have a bunch of siemens gigaset phones with some gigaset ISDN (BRI) centrals, how can we keep using those (ideally, I'd have a device that speaks some voip and where I can register those siemens dect phones with) |
| 14:47.49 | orsonork | i want to play a sound while briding multiple calls and that everyone listen the sound |
| 14:48.17 | [TK]D-Fender | orsonork: option "M()" . go READ. |
| 14:48.32 | orsonork | ok |
| 14:48.41 | *** join/#asterisk e` (n=e@38.102.196.202) |
| 14:49.05 | tzafrir | ronr, there are quite a few ways to connect ISDN BRI to Asterisk |
| 14:49.19 | tzafrir | In fact, many of the simple ISDN PCI cards will do |
| 14:50.22 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
| 14:50.42 | grandpapadot | Is there a way to 'grab' a ringing extension from another phone in 1.2.x? |
| 14:51.04 | *** join/#asterisk fskrotzki (n=fskrot@host198.textwise.com) |
| 14:51.09 | [TK]D-Fender | grandpapadot: Yes. "show applications" <- go find which one |
| 14:51.39 | ronr | tzafrir: I won't be using ISDN BRI, currently the gigasets are connected to three isdn bri pbx, but 1 asterisk server should replace that (with 1 E1 line) |
| 14:51.43 | grandpapadot | Ahh. Pickup. Duh. Thanks. |
| 14:52.18 | tzafrir | ronr, right. Do you need to connect BRI handsets? |
| 14:53.08 | ronr | I need to connect gigaset phone (like gigaset 4000) |
| 14:53.42 | tzafrir | Though 3 BRI lines may still be cheaper than 1 fractional E1. Really not sure of the costs at where you are (or anywhere else, actually) |
| 14:53.48 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 14:53.48 | *** mode/#asterisk [+o anthm] by ChanServ |
| 14:53.57 | ronr | currently, they connect to a gigaset isdn pbx, I wouldn't mind tossing those pbx's out |
| 14:54.06 | tzafrir | A phone, or multiple phones? |
| 14:54.23 | ronr | the e1 gives us 15 lines, the 3 bri 6 lines, we need more than 6 lines (and it's actually cheaper) |
| 14:54.27 | ronr | multiple |
| 14:54.46 | ronr | I guess about 10 |
| 14:57.53 | *** join/#asterisk mindCrime (n=chatzill@66.83.208.218.nw.nuvox.net) |
| 14:59.38 | deeperror | any clues why I would hear a beep or tone on all channels when hanging up any of the other zap channels? |
| 15:01.41 | *** join/#asterisk harpal (n=Harpal@124.125.255.223) |
| 15:04.30 | harpal | Do I need Digium Dev-Lite Kit? I want to connect asterisk with softphones only. |
| 15:04.46 | harpal | becaue I dont have any card to connect PSTN |
| 15:04.47 | *** join/#asterisk Lawbringer (n=Lawbring@cpc4-rdng12-0-0-cust292.winn.cable.ntl.com) |
| 15:05.20 | [TK]D-Fender | harpal: Besides using softphones, what else do you want to do? |
| 15:06.17 | harpal | I just want to know working of asterisk. so I want to just connect two phones and than do talk in it. |
| 15:06.53 | [TK]D-Fender | harpal: Then no, you have no need of any special hardware |
| 15:07.33 | harpal | [TK]D-Fender, Just configure asterisk and connect two softphone right? |
| 15:07.40 | [TK]D-Fender | harpal: Correct |
| 15:07.42 | *** join/#asterisk alrs (n=lars@pozug.com) |
| 15:09.44 | harpal | [TK]D-Fender, Thanks again. |
| 15:13.27 | Greek-Boy | !/bin/bash is the interpreter for scripts in debian, right? |
| 15:13.29 | *** join/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 15:14.05 | nestAr | #!/bin/bash |
| 15:14.11 | tzafrir | Greek-Boy, only if explicitly written as #!/bin/bash |
| 15:14.14 | nestAr | or probably #!/bin/sh |
| 15:14.38 | tzafrir | Greek-Boy, if the interpeter is #!/bin/sh , you should not assume bash-specific features |
| 15:14.41 | shawdog22 | I've got 3 people who check a 'shared' voicemail box. Is there any clever ways of locking while someone is currently logged into it? |
| 15:14.49 | Greek-Boy | i tried running this |
| 15:14.50 | Greek-Boy | lg-asterisk01:/var/spool/asterisk/monitor# /usr/local/bin/2wav2mp3_suse internal-tos--in.wav internal-tos--out.wav try.mp3 |
| 15:14.50 | Greek-Boy | -bash: /usr/local/bin/2wav2mp3_suse: /bin/sh^M: bad interpreter: No such file or directory |
| 15:14.56 | tzafrir | This allows the admin to replace /bin/sh with a lighter shell |
| 15:15.03 | Greek-Boy | in the script it uses #!/bin/sh |
| 15:15.07 | nestAr | shawdog22: you could use a setgroup routine |
| 15:15.13 | bobkare | if bash is run as #!/bin/sh it runs in a special retarded mode where it doesn't support all special bashisms |
| 15:15.23 | tzafrir | Greek-Boy, edit your shell scripts with an editor that knows what unix text files are |
| 15:15.25 | [TK]D-Fender | shawdog22: lock from reading basically? |
| 15:15.40 | Greek-Boy | tzafrir i used vim |
| 15:16.04 | tzafrir | and in vim you see '[dos]' down below, right? |
| 15:16.48 | tzafrir | so in vim run %s/\r$// |
| 15:16.52 | tzafrir | or something similar |
| 15:17.17 | shawdog22 | [TK]D-Fender: Pretty much, if person 1 is logged listening to messages, persons 2 and 3 try and login they get a 'in-use' message, till 1 logs out. |
| 15:18.05 | shawdog22 | [TK]D-Fender: Too many people logging in at the same time, and grabbing the same message. And poor communication on who already has been called back. |
| 15:18.25 | *** join/#asterisk tcseke (n=chatzill@22-36.adsl.etel.hu) |
| 15:19.05 | Greek-Boy | tzafrir i did not see [dos] in vim |
| 15:19.17 | Greek-Boy | and i tried to save the script with vi instead of vim but same result |
| 15:19.50 | [TK]D-Fender | shawdog22: use the GROUP method that nestAr was suggesting |
| 15:20.01 | tzafrir | and you don't see a ^M in the first line in vim? |
| 15:20.15 | tzafrir | anyway, this is getting off-topic for this channel |
| 15:20.29 | Zuchmir | is there any simple way of creatng a menu system similar to http://pastebin.ca/784037 (where not all menus have the same amount of submenus/items)? |
| 15:20.54 | Zuchmir | ... where the depth of the menu can vary |
| 15:21.08 | Greek-Boy | lol tzafrir; u right |
| 15:21.35 | Greek-Boy | dont worry, i'll just rewrite the file in vi from scratch and see what happens |
| 15:21.56 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 15:22.42 | *** join/#asterisk Toerkeium (i=oo@201.216.206.221) |
| 15:23.26 | [TK]D-Fender | Zuchmir: each is in its own context. You can make them as similar or as different as you want |
| 15:24.17 | [TK]D-Fender | Zuchmir: And ControlPlayback doe NOT look lie a valid way to even implement a "menu". |
| 15:24.31 | [TK]D-Fender | Zuchmir: Where do you even allow input? |
| 15:24.35 | shawdog22 | [TK]D-Fender & nestAr: Thanks for the info. |
| 15:25.10 | [TK]D-Fender | Zuchmir: And I've just noticed that it is mis-spelled "ControlPayback". Is this some sort of custom made app? |
| 15:25.45 | Kobaz | [Nov 19 10:24:16] WARNING[21749]: app_meetme.c:772 build_conf: Unable to open pseudo device |
| 15:25.51 | Kobaz | so is that bad? |
| 15:26.00 | [TK]D-Fender | Kobaz: it is if you want Meetme to work |
| 15:26.07 | nestAr | Kobaz: yea, you need ztdummy |
| 15:26.22 | [TK]D-Fender | Kobaz: You have a Zaptel card, or are you using ztdummy? Either way, Zaptel was not loaded before starting * |
| 15:26.29 | *** part/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 15:26.49 | *** join/#asterisk mistik1 (n=mistik1@ool-4352c7d3.dyn.optonline.net) |
| 15:27.00 | mistik1 | hello everyone |
| 15:27.06 | Kobaz | ztdummy is loaded |
| 15:27.21 | [TK]D-Fender | Kobaz: stop *, do "ztcfg -vvvv" and then restart * and tes |
| 15:27.25 | Zuchmir | [TK]D-Fender: that was not a copy/paste from an extensions.conf, that was simply a thoery, and yes you are right, Background() / ControlPlayback() were mixed up in that example |
| 15:27.28 | mistik1 | Is there a comparison chart of the differences between asterisknow and something like FreePBX? |
| 15:27.43 | [TK]D-Fender | Zuchmir: amongst other things. |
| 15:27.46 | Kobaz | k |
| 15:27.57 | [TK]D-Fender | Zuchmir: Very incomplete sample. |
| 15:28.35 | [TK]D-Fender | mistik1: Go check on their sites and respective channels. Niether are supported here. |
| 15:29.21 | Kobaz | mm |
| 15:29.27 | Zuchmir | [TK]D-Fender: what i really want is very complex, and as far as i can see, * will not handle it, but i assume that a simple multi-layered menu would be pretty standard |
| 15:29.29 | Kobaz | i think i need firmware loading support |
| 15:29.32 | *** part/#asterisk bobkare (i=bob@cakebox.net) |
| 15:29.36 | Kobaz | line 0: Unable to open master device '/dev/zap/ctl' |
| 15:30.04 | [TK]D-Fender | Zuchmir: How complex? I haven't seen anything that * cannot handle.... |
| 15:30.33 | *** part/#asterisk mistik1 (n=mistik1@ool-4352c7d3.dyn.optonline.net) |
| 15:30.39 | [TK]D-Fender | Kobaz: not good.... that is an OS module load failure I'm betting. Others can guide you better from here... |
| 15:30.53 | Kobaz | yeah i just built the firmware module |
| 15:31.29 | *** join/#asterisk clandmeter (n=Carlo@81.175.82.2) |
| 15:32.02 | De_Mon | [TK]D-Fender he's after a dynamic dialplan that supports unlimited submenus and levels of submenus. |
| 15:32.30 | [TK]D-Fender | De_Mon: There are ways to do that, and samples on the WIKI |
| 15:33.14 | [TK]D-Fender | De_Mon: And "fairly simple". |
| 15:33.15 | De_Mon | I haven't seen the wiki examples, you talking about the forbidden voip-info wiki? |
| 15:33.32 | [TK]D-Fender | De_Mon: Yes the WIKI, and not "forbidden". |
| 15:33.33 | De_Mon | [TK]D-Fender couldn't agree more |
| 15:33.42 | [TK]D-Fender | De_Mon: would be better worded as "with a grain of salt. |
| 15:33.56 | De_Mon | snickers |
| 15:34.07 | [TK]D-Fender | De_Mon: Perhaps a small BAG of salt.... jsut remember which shoulder to toss it over :p |
| 15:34.09 | *** join/#asterisk Mercestes (n=Merceste@c-76-30-153-176.hsd1.tx.comcast.net) |
| 15:35.09 | Kobaz | oh |
| 15:35.10 | Kobaz | that worked |
| 15:35.21 | Kobaz | i had to mknod /dev/zap/ctl |
| 15:36.10 | *** join/#asterisk FireMac (n=firemac@CPE000d88ae88b9-CM0011ae8bb0ee.cpe.net.cable.rogers.com) |
| 15:36.18 | Mercestes | Ok, get this. I stumbled into this trixbox box with some dialplan issues. heh. Imagine that. |
| 15:36.34 | Mercestes | So I deloused the server, yum remove trixbox, and downloaded all the source, recompled, etc. etc. |
| 15:36.34 | [TK]D-Fender | Mercestes: load chan_duh.so |
| 15:37.10 | Mercestes | and I copied some old configs over, sip.conf, etc, new extensions.conf though. NOthing usable in that trixbox mess.. |
| 15:37.12 | [TK]D-Fender | "To make an apple pie from scratch one must first create the Universe". |
| 15:37.15 | *** join/#asterisk orsonork (n=orsonork@190.128.168.24) |
| 15:37.28 | orsonork | hello |
| 15:37.50 | Mercestes | I have ${TRUNK}= defined as Zap/g1 in globals. When I reboot, sometimes, it tries to dial out as Zap/g2 but if I do my dialplan reload (1.4 for me, yay!), it...fixes it. |
| 15:38.04 | Mercestes | g2 is not referenced anywhere else in my config files anywhere. =/ |
| 15:38.05 | Mercestes | what gives? |
| 15:38.10 | [TK]D-Fender | Mercestes: PASTEBIN <- |
| 15:38.12 | Kobaz | hmmmm |
| 15:38.33 | [TK]D-Fender | Mercestes: And of course you're wrong, its just a question of WHERE :) |
| 15:39.12 | Mercestes | grep g2 *.conf : a2billing.conf and rpt.conf |
| 15:39.22 | Mercestes | winnar = me. |
| 15:39.34 | Zuchmir | [TK]D-Fender: http://pastebin.ca/784117 again this is not .conf code, but an idea of the complexity desired (which we currently have in our existing PBX) |
| 15:41.02 | Kobaz | are there any other devices other than /dev/zap/ctl that meetme needs? |
| 15:41.12 | Kobaz | it still says can't open pseudo device |
| 15:42.23 | [TK]D-Fender | Zuchmir: Read this for some inspiration : http://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu |
| 15:42.36 | [TK]D-Fender | Zuchmir: And it may be worth it for you to use Realtime for this. |
| 15:43.16 | Kobaz | do de do |
| 15:43.22 | De_Mon | Mercestes do dialplan show and pastebin that too |
| 15:43.28 | Mercestes | [TK]D-Fender, I agree tho, I probably have some 3rd party PITA that's installing some secondary configs or some nice statics or something. |
| 15:43.45 | De_Mon | when its trying to dial g2 that is, not now when its working correctly |
| 15:43.48 | Zuchmir | [TK]D-Fender: I was thinking along those lines, but couldn't figure out how to do this (i'd prefer C) |
| 15:43.50 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 15:43.50 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:44.33 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 15:44.39 | [TK]D-Fender | Zuchmir: You could jsut do the whole thing in AGI and use some other structure to regulate your inputs. And why would your structure change all the time? |
| 15:44.47 | Mercestes | grep g2 *.ael |
| 15:44.52 | Mercestes | <PROTECTED> |
| 15:45.17 | Kobaz | [pid 2423] open("/dev/zap/pseudo", O_RDONLY|O_LARGEFILE) = -1 ENOENT (No such file or directory) |
| 15:45.20 | Kobaz | bing |
| 15:45.27 | Mercestes | De_Mon, ding ding ding ding, we have a winnar. |
| 15:45.34 | Kobaz | there is other devices |
| 15:45.55 | De_Mon | Mercestes good one. whos idea was it to use .ael for those config files! ;P |
| 15:46.02 | Mercestes | De_Mon: It's the default. |
| 15:46.15 | Kobaz | is there a script that's supposed to make the /dev/zap devices? |
| 15:46.22 | De_Mon | doesn't it "have" to be .ael though? |
| 15:46.36 | Mercestes | It's from make samples. |
| 15:46.38 | Zuchmir | the structure doesn't change often, but has to have the ability to change |
| 15:47.05 | De_Mon | Zuchmir dialplans arn't written in stone you know |
| 15:47.26 | Mercestes | and the stock configs global trunk=Zap/g2 for some reason. So I guess on a system reboot, ael takes presidence while a dialplan reload gives extensions.conf precidence. |
| 15:47.31 | De_Mon | you just don't want to be the one changing it do you |
| 15:47.49 | De_Mon | Mercestes sounds like a bug! |
| 15:47.51 | Zuchmir | also, i don't want to lose the ff/rewind capability while adding prev/next |
| 15:48.03 | Mercestes | Yay! |
| 15:49.23 | [TK]D-Fender | Zuchmir: Still Doable. |
| 15:49.38 | De_Mon | Zuchmir how do you intent to make redable recording names, without writing them in by hand? |
| 15:49.45 | De_Mon | human-readable |
| 15:49.55 | *** join/#asterisk cjk (n=loic@80.92.64.103) |
| 15:50.24 | cjk | hi, is there a way to disable that asterisk answeres to qualify packages? in fact my box does not answer to them and i would like to enable it |
| 15:51.04 | Mercestes | cjk: qualfiy=yes or qualify=no or qualify=2000 |
| 15:51.31 | cjk | Mercestes, my box2 sends out the packages to box1, but box1 does not answer |
| 15:51.38 | *** join/#asterisk |dennis| (n=dennis@200.32.236.10) |
| 15:52.03 | Mercestes | cjk: Is box1 a sip peer of box2? |
| 15:52.04 | Zuchmir | De_Mon: human readable in the sense that each "series" has it's own folder etc |
| 15:52.19 | cjk | Mercestes, yes |
| 15:53.36 | Kobaz | okay next question |
| 15:53.43 | Kobaz | why would ${CALLERID} be empty |
| 15:53.48 | De_Mon | oh brother. so you didn't like that recordingname${LEVEL} because there wasnt a ${LEVEL}/ in front of it? you just want to do this in C and nothing we tell you will change your mind will it. |
| 15:54.03 | Kobaz | i'm dialing from an iax2 extension |
| 15:54.37 | Mercestes | I hate to admit but D-Fender was right earlier so winnar != me, winnar = D-fender..:( |
| 15:54.39 | tzafrir | Greek-Boy, that ^M is just a character. You can delete it |
| 15:54.43 | Mercestes | cjk: same subnet? |
| 15:54.51 | cjk | Mercestes, no |
| 15:54.56 | tzafrir | Or add another one (ctrl-v ctrl-m) |
| 15:55.00 | cjk | Mercestes, but calls from box1 to box2 are working |
| 15:55.37 | *** join/#asterisk mackes (n=root@65-121-253-83.dia.static.qwest.net) |
| 15:55.40 | Mercestes | cjk: qualify=yes on both boxes? |
| 15:55.58 | cjk | Mercestes, no, why should it? |
| 15:56.01 | [TK]D-Fender | Kobaz: That var is deprecated in 1.2. Go read UPGRADE.TXT and the rest of the documentation in your source DOCS folder |
| 15:56.16 | Mercestes | cjk: make them yes then. |
| 15:56.34 | cjk | Mercestes, i did that too but didnt change anything.... |
| 15:56.53 | Kobaz | [TK]D-Fender: ah |
| 15:57.10 | Kobaz | yeah i was wondering... it used to work |
| 15:57.17 | Mercestes | cjk: make it yes on both boxes and sip reload |
| 15:57.40 | cjk | Mercestes, i tried already |
| 15:57.43 | mackes | Hey Hey Hye |
| 15:58.32 | Mercestes | cjk: I understand, but it's supposed to be yes, so do that, reload, and then verify breakage. |
| 15:58.52 | Zuchmir | De_Mon: i am not bent on any technology, if dialpad can do it easier than C, i'm all for it, but i still can't see how to implement different size menus, ie one menu can have 4 subitems, another can have 53, how do i make sure on a subitem needing 4 the user can press 1 and immediatly get file, and yet in the 53 option menu all accessible |
| 15:59.07 | cjk | Mercestes, thanks, i will try to figure out |
| 15:59.40 | Zuchmir | ... i'm reading through that wiki page, see if that helps... |
| 16:00.26 | Mercestes | Zuchmir, You can do that through contexts, if I'm understanding you correctly. |
| 16:00.36 | De_Mon | Zuchmir with patern matching _XX will match any 2 digit number, and ${EXTEN} is the number they pressed. |
| 16:01.50 | jameswf | jbot, rose |
| 16:01.57 | Zuchmir | yeah, but _XX wants 2 digits, and will have to "timeout" on "1" vs "10" |
| 16:02.41 | De_Mon | yes, digit timeout determins how long it will wait |
| 16:02.44 | Mercestes | Only like 3 seconds. |
| 16:02.48 | *** join/#asterisk Klydal (n=Klydal@r74-192-234-206.lfkncmta01.lfkntx.tl.dh.suddenlink.net) |
| 16:02.50 | Zuchmir | ...and when there's only 4 items, the user expect an instant response on a singal digit |
| 16:02.50 | *** join/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 16:02.54 | Mercestes | or 1 second if you have fast dialers. |
| 16:03.11 | De_Mon | that problem exists no matter how you create the extensions |
| 16:03.24 | Mercestes | Unless you use different contexts, one only matches one digit adn the other matches 2. |
| 16:03.46 | Mercestes | but then youhave a "press 1 for 4 options, press 2 for 53 options." type of deal. |
| 16:03.51 | De_Mon | if there is a 10 and a 1, how is asterisk supposed to know if the user plans to press another number. chan_telepathy isn't finished yet. |
| 16:04.18 | Zuchmir | if you have detailed extentions you can do _[0-4] ... vs _XX |
| 16:04.22 | JayTee52 | ah, but when chan_telepathy is finished you'll never have to dial your phone again. It will call you. |
| 16:04.42 | Mercestes | Yes, but you still have the problem with 1 vs 10-19. |
| 16:04.43 | De_Mon | oh that, true enough. AGI or realtime then. |
| 16:04.54 | shawdog22 | [TK]D-Fender: I did some reading on the set GROUP, for my multiple user voicemail issue. And I'm a little confused on how to monitor which mailbox is being accessed. |
| 16:05.25 | [TK]D-Fender | shawdog22: Set the group based on the box# |
| 16:05.39 | Klydal | Ok, I've come to ask the experts (hopefully you guys) which wifi phone I should get. Im looking to buy some for my family as a christmas present. Price is probably number 1 priority. I am looking at Utstarcom F1000 and the Linksys wip300. Any suggestions? Feel free to pm me if you like. |
| 16:05.42 | De_Mon | speaking of extensions, why does dialing 1# make * think I want to call extension '1#' I just wanted it to stop waiting for digits! |
| 16:05.48 | Mercestes | ~wifi |
| 16:05.49 | jbot | somebody said wifi was see wireless or for a small compact non-port-blocking card, get one of these a) linksys wcf12 for only $65 shipped from buy.com b) netgear MA701NA for $65 shipped from buy.com c) socket LOW POWER wlan (amazing battery life) for $160 + shipping on buy.com, or better than nothing |
| 16:06.12 | Mercestes | Klydal, none of them. All wifi phones blow. |
| 16:06.14 | Klydal | oh wow |
| 16:06.22 | De_Mon | yes sucky sucky |
| 16:06.25 | jameswf | wifi phones kinda suck a$$ |
| 16:06.27 | Klydal | sucky how? |
| 16:06.28 | Mercestes | Get a cordless phone with a sip base station or use an ATA on a cordless phone. |
| 16:06.44 | Klydal | that is too much for my other family members |
| 16:06.52 | jameswf | wifi phones are like ummmm cingular.... |
| 16:06.58 | Klydal | haha |
| 16:07.05 | Mercestes | Klydal, sucky as in nervous asian with a teeth grinding tic sucky. |
| 16:07.08 | Mercestes | not at all in a good way. |
| 16:07.19 | shawdog22 | [TK]D-Fender: Will it work if people just dial 400, to access the VoiceMailMain()? |
| 16:07.38 | Mercestes | It works great....for about 10 feet away from your base station...and that's if you use the $200 "long range" antennae. |
| 16:07.38 | [TK]D-Fender | shawdog22: It'll work if you are TELLING VMM what box to enter FIRST. |
| 16:07.55 | shawdog22 | [TK]D-Fender: Yeah, that is what I was afraid of. |
| 16:08.45 | shawdog22 | [TK]D-Fender: People just log into VoiceMailMain and enter their extension. Old dogs new tricks... |
| 16:09.00 | [TK]D-Fender | shawdog22: Go change your dialplan. |
| 16:09.30 | *** join/#asterisk myiagy (n=myiagy@200.215.59.112) |
| 16:09.37 | Mercestes | shawdog22, Could use a ton of voicemail contexts. >.> |
| 16:10.39 | shawdog22 | [TK]D-Fender: I'm one of those guys that is forced to try and replicate an old phone system, due to working with people who refuse to change. |
| 16:11.45 | Mercestes | shawdog22, offer to charge them the same rates that their old phonesystem cost them. |
| 16:11.46 | nestAr | yeah, i had people like that, i told them they could just give up, or they could change.. |
| 16:11.48 | Darthclue | shawdog22, convince them to update...press the button that says messages or dial 400. enter your pass when prompted. |
| 16:11.54 | nestAr | they changed. |
| 16:12.31 | Mercestes | that's always my response..."My old phone system didn't work that way!" "My system costs 1/10th of what yours did. If you want to pay me that much, I'll make it work that way." |
| 16:13.38 | shawdog22 | Ha.. I don't know if it's a unwillingness to change, or the fact of limited brain capacity. |
| 16:13.52 | Mercestes | necessity is the mother of invention. |
| 16:13.52 | nestAr | it's unwillingness |
| 16:14.02 | Darthclue | offer a 'memory upgrade' |
| 16:14.07 | nestAr | they go to a different company, likely they'd be forced to change. |
| 16:14.13 | Kobaz | doing a VoiceMailMain(3000) should just ask for the password for the mailbox... right? |
| 16:14.16 | Darthclue | and don't forget to throw in the extra 'storage capacity' |
| 16:14.29 | Mercestes | Kobaz: If 3000 exists in context default, then yes. |
| 16:14.48 | shawdog22 | Heck, forget upgrades. I'll just buy newer models. |
| 16:14.50 | Kobaz | yeah, it does |
| 16:14.54 | Kobaz | it asks for a mailbox number |
| 16:14.55 | Kobaz | hmm |
| 16:15.21 | nestAr | Kobaz: check your console, should give you a notice of why 3000 wasn't loaded. |
| 16:15.29 | De_Mon | Mercestes I love that line, I'm gonna use it |
| 16:15.35 | Mercestes | :D |
| 16:16.05 | Zuchmir | thanks for your help, i got to get some sleep |
| 16:16.13 | Mercestes | l8s, Zuchmir. |
| 16:16.40 | shawdog22 | Alright, thanks for the info guys. I'll set those features up, and tell them they are available if people want them, and it's up to them to use it. |
| 16:17.45 | shawdog22 | Later-- |
| 16:18.16 | *** part/#asterisk shawdog22 (n=shaw@pacman.oaklandcorp.com) |
| 16:19.06 | Kobaz | nestAr: the extension gets loaded just fine |
| 16:19.18 | Kobaz | nestAr: there's no errors when voicemailmain is running either |
| 16:19.38 | [TK]D-Fender | Kobaz: less talk, more pastebin.... |
| 16:22.25 | Kobaz | http://www.pastebin.ca/784194 |
| 16:23.01 | Kobaz | exten => 3401,1,VoiceMailMain(3000) |
| 16:23.07 | [TK]D-Fender | Kobaz: So it asked for that MB and they entered NOTHING for the pass. |
| 16:23.19 | Kobaz | it's not supposed to ask for mailbox |
| 16:23.25 | Kobaz | it's supposed to just ask for the password |
| 16:23.34 | Kobaz | and yeah i did enter nothing since i've been testingf |
| 16:23.35 | [TK]D-Fender | Kobaz: Its not asking for the box, its asking for the PASSWORD |
| 16:23.55 | [TK]D-Fender | Kobaz: Nobody said thats how to bypass BOTH <---- |
| 16:24.03 | Kobaz | i'm not trying to bypass both |
| 16:24.04 | *** join/#asterisk anonymouz666 (n=anonymou@201.19.134.26) |
| 16:24.14 | Kobaz | i dial 3401... it asks "mailbox" |
| 16:24.15 | [TK]D-Fender | Kobaz: Well you have to give it the PW... |
| 16:24.16 | Kobaz | i enter "3000" |
| 16:24.20 | Kobaz | and then it says "password" |
| 16:24.23 | Kobaz | and then i type 1234 |
| 16:24.25 | Kobaz | and then it lets me in |
| 16:25.02 | [TK]D-Fender | Kobaz: And pastebin your voicemail.conf as well. |
| 16:25.21 | twisted | Kobaz: this sounds extremely fundamental, but have you issued an "extensions reload" since you updated your extensions.conf? |
| 16:25.26 | Kobaz | yeah |
| 16:25.28 | Kobaz | it's been reloaded |
| 16:25.32 | [TK]D-Fender | Kobaz: So far it should only be asking for the PW, which is exactly what it seems to be doing |
| 16:25.33 | twisted | ok, just checking :P |
| 16:25.42 | Kobaz | [TK]D-Fender: it's not doing that though |
| 16:26.05 | [TK]D-Fender | Kobaz: Nowhere in your PB do I see you entering the box # |
| 16:26.40 | Kobaz | i didn't since i'm just calling and testing |
| 16:26.45 | Kobaz | to see if it asks for a password |
| 16:26.49 | Kobaz | i'll paste a whole attempt |
| 16:26.53 | [TK]D-Fender | ....... |
| 16:27.19 | Kobaz | aughh, vpn is dieing... sec |
| 16:27.19 | twisted | [TK]D-Fender: in his PB it clearly shows voicemailmain being executed on box 3000, and the asking for login (which is the "mailbox?" prompt |
| 16:27.39 | Kobaz | it asks for login, then password |
| 16:27.54 | Kobaz | what i want to do after i get this going, is just use the callerid as the box# |
| 16:27.59 | twisted | Kobaz: try putting the @context at the end of the 3000 |
| 16:28.05 | Kobaz | but i threw 3000 in there for sanity |
| 16:28.07 | [TK]D-Fender | twisted: Shouldn't we see the input result of the box? |
| 16:28.12 | twisted | so VoiceMailMain(3000@somecontext) |
| 16:28.14 | Kobaz | yeah i tried that too |
| 16:28.28 | Klydal | so do you guys have any suggestions on a decent ATA? Sipura SPA-1001? |
| 16:28.35 | twisted | [TK]D-Fender: in his PB it shoudl not have even asked for that |
| 16:28.37 | [TK]D-Fender | Kobaz: Please pastebin a better attempt and your configs. No more running in circles... file is already doing that... |
| 16:28.51 | twisted | when you specify a box # in voicemailmain() it should NOT ask you for the box # when you call it |
| 16:28.53 | Kobaz | yeah, i'll do that when the vpn is back up |
| 16:28.58 | Kobaz | twisted: yeap, that's what i said |
| 16:29.02 | twisted | it should ONLY ask for the password. |
| 16:29.05 | [TK]D-Fender | Klydal: That'd me Linksys now, and sure, its decent. |
| 16:29.07 | Kobaz | yeap, but it's not |
| 16:29.07 | Kobaz | heh |
| 16:29.08 | twisted | Kobaz: i know, i'm telling [TK]D-Fender |
| 16:29.10 | Kobaz | yeah |
| 16:29.39 | twisted | regardless if he's entering the box # when it asks for it, the dialplan logic dictates it should not be even asking for it |
| 16:29.44 | Klydal | yeah thats right.. I forgot about that. I have a handytone atm. But im just looking cheap if Im going to get that and a cordless phone |
| 16:30.03 | Mercestes | ~cheap |
| 16:30.04 | jbot | hmm... cheap is a bad idea. If you're setting up an Asterisk system, one of the wisest pieces of advice you can take is this: DON'T BE A CHEAPSKATE. |
| 16:30.14 | Klydal | haha |
| 16:30.18 | Mercestes | :D |
| 16:30.39 | Klydal | yeah, I was cheap and put asterisk on my pc with vmware :P |
| 16:30.51 | Mercestes | Didn't work, did it? |
| 16:30.58 | Klydal | it worked ok |
| 16:31.07 | Klydal | it was an older model ibm laptop |
| 16:31.23 | Mercestes | Until you tried to use meetme or MOH or load up ztdummy |
| 16:31.35 | blitzrage | twisted: unless the voicemail box # doesn't exist |
| 16:31.46 | blitzrage | twisted: and hi! |
| 16:31.54 | mackes | WiFi Phones Baby |
| 16:32.00 | blitzrage | gross |
| 16:32.07 | mackes | The Hitachi WiFi Rocks |
| 16:32.14 | twisted | blitzrage: sure, but if he enters the same box #, and then the password, and it logs in, then it obviously exists. |
| 16:32.18 | twisted | blitzrage: and hi! |
| 16:32.19 | twisted | :P |
| 16:32.27 | blitzrage | twisted: oh ya -- then I agree :) |
| 16:33.01 | Mercestes | Klydal, thats ok, I run asterisk on my Linksys wrt54gl wireless router. |
| 16:33.37 | Klydal | oh nice |
| 16:33.42 | Mercestes | Sorta... |
| 16:33.58 | Mercestes | If by "work" you mean I get a green ok when I do an /etc/init.d/asterisk start, then, yea, it works fine |
| 16:34.09 | Mercestes | if by "work" you mean, I can do more than a single call on it then not so much. |
| 16:34.25 | Klydal | well it was a cool idea |
| 16:34.56 | Mercestes | We used to use it for demos. For some reason, people liked calling each other from 2 polycoms glued to a piece of plywood with a linksys router in the middle. |
| 16:35.16 | Mercestes | It's like, "WOOHOO! I'm on voip!" |
| 16:36.52 | *** join/#asterisk grEvenX (n=even@pc107-130.ktv.no) |
| 16:38.07 | *** join/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 16:39.31 | *** join/#asterisk qdk (n=qdk@0x573fe57e.bynqu2.broadband.tele.dk) |
| 16:43.20 | Kobaz | anyone use polycom 330's? |
| 16:43.30 | Kobaz | if i dial a number outside the dialplan the phone locks up |
| 16:43.40 | Kobaz | well, outside the digit map |
| 16:44.02 | Kobaz | and i fixed the voicemail |
| 16:44.16 | Kobaz | i forgot asterisk needs a hard restart to load in the new voicemail stuff |
| 16:44.33 | Klydal | anyone know how the DECT voip phones are? |
| 16:44.33 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:44.44 | Mercestes | Kobaz, define "locks up" |
| 16:45.03 | Kobaz | well like a pc lockup... the interface completely freezes |
| 16:45.06 | Mercestes | Do you mean "refuses to dial out" or "refuses to accept further user input until I powercycle the thing?" |
| 16:45.11 | Kobaz | and 30 seconds later it reboots |
| 16:45.15 | Mercestes | NICE. |
| 16:45.26 | Mercestes | Sounds like a good time to call your supplier. |
| 16:45.28 | Kobaz | the 501's dont do that |
| 16:45.32 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 16:45.32 | Kobaz | yeah |
| 16:45.54 | Kobaz | so if i do 123 and hit line 1 |
| 16:45.56 | Kobaz | phone dies |
| 16:45.57 | Kobaz | reboots |
| 16:45.59 | *** join/#asterisk CunningPike (n=arodgers@204.239.12.183) |
| 16:46.11 | Kobaz | or anything that's not in the digit map |
| 16:46.13 | Mercestes | Is DigitMapPatternMatching=2 before yoru Digitmap designation? |
| 16:46.18 | Kobaz | mmm |
| 16:47.02 | Mercestes | Oh, that reminds me. Fender!!!!! |
| 16:47.07 | Mercestes | Guess who I had an interview with the other day.... |
| 16:47.12 | Kobaz | nope, i don't have that |
| 16:47.29 | Mercestes | Kobaz, Actually, I think it's called "ImpossibleMatchHandling" or something. |
| 16:47.36 | Kobaz | k |
| 16:47.44 | Mercestes | Kobaz, it's the line before yoru digitmap. set it to 2 and rebootage and see if that's a work around. |
| 16:47.51 | *** join/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 16:47.54 | Kobaz | fgimpossibleMatchHandling="0" |
| 16:47.58 | Kobaz | impossibleMatchHandling="0" |
| 16:48.16 | drmessano | Any reason why srvlookup defaults to no? |
| 16:48.17 | Mercestes | Yea, set it to 2 and see if that magically fixes it. |
| 16:49.42 | Kobaz | waiting for notworking to init |
| 16:49.48 | *** join/#asterisk klictel (n=klictel@atelka.info) |
| 16:49.51 | Kobaz | do de do |
| 16:50.58 | Mercestes | [TK]D-Fender: Guess who I had an interview with the other day. |
| 16:51.13 | *** join/#asterisk krondorl (n=chatzill@207.245.216.9) |
| 16:51.22 | krondorl | Greetings all... |
| 16:51.37 | Mercestes | Greetin's. |
| 16:51.37 | Kobaz | didn't fix it |
| 16:51.48 | Mercestes | Kobaz, nice. Yea, refer to supplier. |
| 16:52.06 | Kobaz | 10 seconds to reboot |
| 16:52.31 | krondorl | Question: How can one get app_amd.c into Asterisk with the gentoo distro that automatically compiles the code? |
| 16:52.47 | krondorl | Version 1.2.17.. |
| 16:53.17 | [TK]D-Fender | Mercestes: Who? |
| 16:53.28 | Mercestes | [TK]D-Fender, Polycom! :D |
| 16:53.35 | *** join/#asterisk oej (n=olle@soll4-125.cust.blixtvik.net) |
| 16:53.41 | [TK]D-Fender | Mercestes: very cool |
| 16:53.49 | [TK]D-Fender | Mercestes: What kind of position? |
| 16:54.04 | Mercestes | krondorl, you can tar -zxvf the source under /usr/portage/sourcefiles or something and edit what you need to edit then manually recompile it. |
| 16:54.09 | Mercestes | [TK]D-Fender, field support for video conferencing. |
| 16:54.14 | tzafrir | krondorl, you should start by finding app_amd for 1.2 |
| 16:54.39 | tzafrir | krondorl, I also hear there's some unofficial asterisk 1.4 for Gentoo |
| 16:55.06 | Mercestes | it's in the voip overlay under layman. |
| 16:55.24 | Mercestes | maintained by the #gentoo-voip folks |
| 16:57.03 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 16:57.03 | *** mode/#asterisk [+o codefreeze] by ChanServ |
| 16:58.18 | Mercestes | [TK]D-Fender, lots of travel, but I get to work from home a lot too and I get to do video conferencing. :D |
| 16:59.02 | [TK]D-Fender | Mercestes: No more cheesy web-camming pr0n for you! Big league! |
| 16:59.21 | Mercestes | Exactly! NOw it's high class web-camming pr0n! |
| 17:00.01 | Mercestes | :D |
| 17:00.14 | jameswf | well tzafrir I made a post on the tb forums about fonality cripling genzaptelconf.... now I will sit back and wait for the "whacho talkin bout willis" emails |
| 17:00.25 | drmessano | lol |
| 17:00.49 | Mercestes | now if I can just not screw up my interviews by randomly mentioning BDSM midget horseporn, I'll be ok... |
| 17:01.28 | jameswf | we would probably hire you :) |
| 17:01.38 | tzafrir | jameswf, Understanding the behaviour of Fonality with Trixbox CE is now beyond me |
| 17:02.13 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-68-93-83-7.dsl.hstntx.swbell.net) |
| 17:02.18 | drmessano | In order to make any product your own, you must break things familiar to others in a way only sensible to yourself |
| 17:02.39 | *** join/#asterisk seba_soy (n=chatzill@190.2.63.135) |
| 17:02.47 | Mercestes | jameswf, If I mentioned the horseporn? Heh, I like your company. |
| 17:02.48 | drmessano | If anyone uses that, I get $5 per use |
| 17:02.55 | seba_soy | Hi |
| 17:03.03 | jameswf | I was going to write a book then realized I am to smart to limit myself to a couple hundred pages |
| 17:03.18 | seba_soy | I have a question maybe someone can help me |
| 17:03.40 | Mercestes | seba_soy, out with it already. |
| 17:04.00 | drmessano | You could always write another Asterisk book |
| 17:04.26 | seba_soy | when I place a call from my phone to a local extension, for example from 101 to 102 I see it uses Macro(stdexten) but I CAN'T FIND where is in extensions.conf this dialplan... |
| 17:04.35 | jameswf | I should do asterisk for dummies. |
| 17:04.43 | Mercestes | litererotica: Asterisk: telephony gone wild. |
| 17:05.06 | Mercestes | seba_soy, grep macro-std-exten *.conf |
| 17:05.10 | jameswf | how to make asterisk a 2 teribyte porn server ISBN:0000 |
| 17:05.17 | Darthclue | Welcome to the Dark Side : The Truth about VOIP (Video Over IP) |
| 17:05.21 | Mercestes | seba_soy, I am certain it is either in extensions.conf, extensions_additional.conf, or in one of it's includes. |
| 17:05.36 | seba_soy | checking.... |
| 17:05.44 | drmessano | "How I stuck it to the man (mom) and made $10000 in free phone calls with Asterisk" by Trey Jones |
| 17:05.46 | jameswf | have festival read dirty stories |
| 17:05.48 | coppice | Porn Serving for Dummies |
| 17:06.23 | Mercestes | Today's book, "Ranch of my dreams" as read by Steven Hawkings. |
| 17:06.47 | [TK]D-Fender | IDC : "Quantum Mechanics for Dummies" <---- |
| 17:07.11 | Mercestes | lol |
| 17:07.13 | drmessano | "Solving One-Way Audio problems in Asterisk" by Fiar Wall |
| 17:07.22 | Mercestes | rofl |
| 17:07.30 | _x86_ | Fiar Wall omg roflmao |
| 17:07.49 | coppice | and the one for new parents "Shopping for Dummies" |
| 17:08.00 | _x86_ | coppice you lose |
| 17:08.08 | seba_soy | Mercestes: nothing, there is nothing ... :(... I can't understand how it works... |
| 17:08.21 | jameswf | you know i solved oneway audio by dumping sip for iax |
| 17:08.32 | Mercestes | seba_soy, oh, ya know what? I screwed that one up. lol |
| 17:08.42 | Mercestes | seba_soy, grep macro-stdexten *.conf instead. |
| 17:08.43 | [TK]D-Fender | seba_soy: go look in all of your configs, including AEL junk... |
| 17:08.50 | drmessano | "How I got my Asterisk based PBX out of my LAN" by Nat Conf |
| 17:08.59 | Mercestes | seba_soy, too many -'s there. I always thought it should be std-exten |
| 17:09.14 | jameswf | "Let your kids eat paint chips and save the world" by natural selection |
| 17:09.18 | seba_soy | the Macro is there |
| 17:09.20 | drmessano | lol |
| 17:09.22 | krondorl | tzafrir: Can't use 1.4 series.. Dialer not working well with it.. |
| 17:09.30 | seba_soy | I cant find WHEN Asterisk call this macro |
| 17:09.30 | Mercestes | rob0, ...? |
| 17:09.47 | seba_soy | something like _XXX,1,Macro(stdexten.... |
| 17:09.51 | jameswf | jbot, dropdatabase; |
| 17:09.52 | jbot | So you think you could kill me, did you. You thought you could come in here, in my own home, and take me down without a fight. I would have given you a chance, really. But now, you have unleashed hell. May god have mercy on your soul. |
| 17:10.02 | Mercestes | seba_soy, oh, when? When you dial an extension. Look in the context under extensions.conf, the context beign whatever context= in sip.conf. |
| 17:10.10 | [TK]D-Fender | seba_soy: enable channel debug and you'll see exactly.... |
| 17:10.20 | seba_soy | that's what I looking and I cant find... |
| 17:10.21 | drmessano | "How I would have done it this time, since I didn't do it last time, and not this time either" by OJ Simpson |
| 17:10.34 | seba_soy | some change using asterisk GUI? |
| 17:10.39 | seba_soy | I confiure all using the GUI |
| 17:10.45 | jameswf | property recovery by OJ simpson |
| 17:10.49 | waKKu | jbot i know that phrase... where is it from ? :) |
| 17:10.50 | jbot | You know that phrase... where is it from ? :)? |
| 17:10.50 | drmessano | lol |
| 17:11.02 | waKKu | ¬¬ |
| 17:11.06 | Mercestes | rob0, I wasn't aware Shakespeare wrote a play entitled "Gay Boys in Bondage." |
| 17:11.23 | coppice | its the Bard's latest work |
| 17:11.29 | jameswf | jbot, porn |
| 17:11.30 | jbot | Porn remains one of the largest problems with Open Source Software. Often causing development delays, flooded links and, in extreme cases, disabling programmers ability to type. |
| 17:11.44 | anonymouz666 | hahaha |
| 17:11.45 | drmessano | "How OJ got my Asterisk server back" by Stohlen Peebex |
| 17:12.00 | *** join/#asterisk Jameno123 (n=james@63.210.246.34) |
| 17:12.30 | Darthclue | "ET Phone Home : Lowjack for your PBX" |
| 17:12.35 | Jameno123 | Which module do you load for the TE120P |
| 17:12.42 | *** join/#asterisk key2 (n=Ritual@193.33.36.20) |
| 17:12.45 | Jameno123 | wct1xxp or wcte11xp |
| 17:12.48 | cpm | Mercestes, Oh ...Gay Boys in Bondage' What, izzit- tragedy? Comedy? |
| 17:12.49 | krondorl | DANG.. No one is responding to my question in gentoo-voip... |
| 17:12.56 | jameswf | Micheal Jackson: How to build a small footprint asterisk PBX and mastering the touch command |
| 17:12.59 | Mercestes | cpm: I'm guessing tragedy. |
| 17:13.14 | cpm | hint |
| 17:13.15 | krondorl | cpm: I'm guessing horror |
| 17:13.16 | cpm | 'tis a story of a man's great love for his... fellow men. |
| 17:13.33 | jameswf | brokeback Asterisk: a love story |
| 17:13.37 | Mercestes | no, Rob0 clearly specified boys. |
| 17:13.41 | cpm | heh |
| 17:13.47 | drmessano | "Asterisktile Dysfunction: Why women really dislike your X100P" |
| 17:13.58 | Mercestes | I'm thinking...that all the first born or second born (if the first ones are too old) get kidnapped... |
| 17:14.05 | Mercestes | and the hero goes to rescue them.... |
| 17:14.49 | coppice | hasn't anyone heer ever studied the classics? |
| 17:14.49 | Mercestes | and then he finds them all tied up, and finds out, that he has a thing for bondage *and* gay sex all at the same time... |
| 17:14.49 | Mercestes | so he does them all, then is afraid that they will tell, so he kills them all. |
| 17:14.49 | jameswf | Viagra Module: the secret to long server uptimes and quick server response |
| 17:14.49 | coppice | rob0: we seem to be amongst heathens |
| 17:14.49 | Mercestes | then the guards show up, and see a bunch of raped massacred boys, so they kill the hero. |
| 17:15.03 | jameswf | jbot, Viagra |
| 17:15.04 | jbot | [viagra] the nickname for the Woody Tech Support Crew |
| 17:15.07 | coppice | rob0: we seem to be amongst *sicko* heathens |
| 17:15.16 | cpm | Now good wife, while I rest, read to me a while from Shakespeare's Gay boys in bondage |
| 17:15.29 | Mercestes | then the french show up, and see the dead boys and the dead hero, and mistake the scene..and slaughter the guards. |
| 17:15.31 | coppice | ah, a cultured man |
| 17:15.35 | drmessano | "He never called me back *SOB*: a womans story of Zaptel issues" |
| 17:15.39 | *** join/#asterisk |omni| (n=rob@c-67-185-56-106.hsd1.wa.comcast.net) |
| 17:15.40 | key2 | who is the dude that wrote idefisk/zoiper ? |
| 17:15.53 | jameswf | zoiper is kinda ass |
| 17:16.20 | key2 | jameswf: what did you code ? |
| 17:16.45 | _charly_ | which iax softphone would you recommend? |
| 17:16.46 | Mercestes | HPEC for Dummies, by Wai Yu Echo. |
| 17:16.51 | jameswf | key2, today? |
| 17:17.00 | Jameno123 | TE120P == which module: wct1xxp or wcte11xp? |
| 17:17.04 | jameswf | I kinda like moxphone |
| 17:17.14 | jameswf | *mozphone |
| 17:17.27 | *** join/#asterisk sigmounte (n=sigmount@88.172.80.96) |
| 17:17.32 | cpm | Dirty books, please. |
| 17:17.34 | jameswf | no fluff doesnt hog memory |
| 17:17.35 | sigmounte | hello any guru about ztdummy ? |
| 17:17.45 | jameswf | dummy guru |
| 17:17.47 | drmessano | "How to create your own Asterisk PBX distro in 7237576 easy steps and profit" |
| 17:18.02 | Mercestes | sigmounte, PRI or Analog? |
| 17:18.17 | jameswf | Hybrid Hosted Solutions: The Future of Technology |
| 17:18.24 | drmessano | lol |
| 17:18.32 | jameswf | *telephony |
| 17:18.41 | sigmounte | i don't have any zaptel card , so to use meetme i've read i have to load ztdummy , but i have no sound and rtc error filling my syslog :( |
| 17:18.46 | drmessano | "Where the hell is my server?: One mans hosted PBX story" |
| 17:18.52 | Jameno123 | nm the sysconfig/zaptel seems out of date :( i see wcte12xp -- nm. |
| 17:19.09 | jameswf | sigmounte, rtc is your board not asterisk |
| 17:19.11 | Mercestes | sigmounte, did you modprobe ztdummy? |
| 17:20.11 | sigmounte | Mercestes, i've use modconf to load ztdummy , same has modprobe , no ? Jameno123 i know , ztdummy use the rtc of the board to provid timing for the meetme app instead of the zaptel card |
| 17:20.33 | jameswf | How I removed a rootkit and killed a call center: the trixbox pro eeceecedredr story |
| 17:20.44 | drmessano | hahahah |
| 17:20.47 | [TK]D-Fender | cpm>Dirty books, please. <- "Hole in the Mattress", Mr. Completely. |
| 17:21.47 | *** join/#asterisk Strom_M (n=strom@ip70-170-60-8.lv.lv.cox.net) |
| 17:22.07 | cpm | aiiee! |
| 17:22.24 | Maxxed | how do i go about booting linux from grub, but having it ignore loading a module? |
| 17:22.38 | drmessano | Need one for admins that upgrade to much |
| 17:22.44 | drmessano | too* |
| 17:23.26 | drmessano | Never heard of someone being accused of upgrading too much, especially when the product has existing issues |
| 17:23.44 | *** join/#asterisk macros73 (n=cs@c-24-3-246-27.hsd1.mn.comcast.net) |
| 17:23.45 | peanut- | I have |
| 17:23.59 | [TK]D-Fender | cpm> "Chinese Child Pornography", Wii Fukem Yeoung |
| 17:24.02 | peanut- | IOS, you don't upgrade unless you absolutely need to |
| 17:24.03 | drmessano | oh? |
| 17:24.12 | *** join/#asterisk russellb (i=russell@asterisk/developer-and-stable-maintainer/drumkilla) |
| 17:24.12 | *** mode/#asterisk [+o russellb] by ChanServ |
| 17:24.30 | drmessano | Yes, but are there enough IOS issues to cause you to WANT to upgrade? |
| 17:24.44 | peanut- | yes |
| 17:24.46 | cpm | http://www.youtube.com/watch?v=v36MCcRPRTc |
| 17:25.47 | drmessano | HA |
| 17:25.50 | drmessano | FIAR!!! |
| 17:26.49 | coppice | beyond turbo - afterburners :-) |
| 17:26.53 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-574b9a75e6d08d56) |
| 17:26.53 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 17:27.24 | drmessano | Makes me want to rent "Top Gun" |
| 17:30.59 | nestAr | i wish they still shot flames like that. |
| 17:32.24 | nestAr | my old car used to shoot fire balls |
| 17:32.31 | nestAr | but not continuous flames |
| 17:34.51 | drmessano | my old car shot water out the back |
| 17:35.07 | drmessano | Trick is to put the crack in the block in JUST the right place |
| 17:41.46 | Mercestes | my go-kart used to shoot flames. |
| 17:41.54 | *** join/#asterisk bantu (n=Miranda@p54A32AD1.dip0.t-ipconnect.de) |
| 17:41.55 | Mercestes | but that's far less impressive. =/ |
| 17:42.40 | [TK]D-Fender | Mercestes: Just move it further up to the cabin ;) |
| 17:42.50 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 17:42.51 | [TK]D-Fender | Mercestes: Burning Man!@ |
| 17:43.22 | coppice | nothing matches the sight and sound of a Saab Viggen on afterburners :-\ |
| 17:43.48 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 17:44.47 | outtolunc | that dude on youtube laying on the bed with a lighter is probably a close second <G> |
| 17:47.53 | [TK]D-Fender | Mercestes: http://www.youtube.com/watch?v=ynZxVErTovg <-- skip to 2:00 :) |
| 17:48.28 | *** join/#asterisk km- (n=pgrace@aeneas.fierymoon.com) |
| 17:48.47 | km- | I'm having a brainfart -- how can I tell which codec a particular sip call is using? show call (sipid) doesn't seem to have it. |
| 17:49.13 | [TK]D-Fender | km-: "sip show channels" |
| 17:49.21 | [TK]D-Fender | km-: "sip show channel [channel]" |
| 17:49.29 | km- | ohhhh I have to preface with sip |
| 17:49.30 | km- | got it, thanks |
| 17:50.03 | *** join/#asterisk galeras (n=galeras@200.31.204.42) |
| 17:51.36 | drmessano | so, srvlookup defaulting to "no" any reason for that? |
| 17:51.57 | galeras | Dear Sirs, Which ring strategy do you recomend to assign an equitable quantity of calls to all agents ? |
| 17:52.18 | galeras | random, rrmemory? |
| 17:52.36 | coppice | [TK]D-Fender: what were those guys trying to achieve with the concrete mixer? |
| 17:53.24 | *** join/#asterisk jmesquita (n=jmesquit@201.7.117.114) |
| 17:53.47 | [TK]D-Fender | coppice: this was at the end of some other "is reality anything like the movies?" Myths, and that one wasn't So in the end, they wanted a "big bang", so they MADE one. This was to highlight the differences between Hollywood explosions and true high-explosives. |
| 17:54.00 | [TK]D-Fender | galeras: rrmemory. |
| 17:54.34 | coppice | dunno. it looked absurd enough to be a hollywood job :-\ |
| 17:55.53 | outtolunc | they were just trying to get the dried concrete out <G> |
| 17:56.06 | outtolunc | sheesh <G> |
| 17:56.22 | *** join/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 17:56.48 | coppice | i don't think they were really trying |
| 17:57.31 | coppice | I saw someone clean one the more conventional way, with a pneumatic drill. that's gotta be one of the nastiest jobs i've ever seen |
| 17:58.00 | waverly360 | Hey guys, if I wanted to get someone up to speed on the basics of configuring and troubleshooting asterisk, are there any classes or training seminars any of you would suggest? I'm looking at the asterisk bootcamp class currently, and I want to send a couple of people there. I don't want to do it if it's not worthwhile though. |
| 17:58.21 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 18:00.11 | *** part/#asterisk galeras (n=galeras@200.31.204.42) |
| 18:02.04 | *** join/#asterisk heliosj (n=jeff@pdpc/supporter/active/xheliox) |
| 18:05.37 | [TK]D-Fender | waverly360: probably about as good a place as you'll find |
| 18:05.53 | *** join/#asterisk jmls (n=jmls@62.49.235.130) |
| 18:05.58 | jmls | evening all |
| 18:07.54 | jmls | can you get the queue information (show queue xxx) from the dialplan using an application or function ? |
| 18:07.54 | drmessano | I only buy dummies books |
| 18:08.06 | drmessano | I feel like they wrote them for me :( |
| 18:08.34 | drmessano | Everytime I see "IAX" spelled out "I A X", I think "Dad?" |
| 18:10.04 | [TK]D-Fender | jmls: nope, parsing time in AGI.... |
| 18:10.09 | waverly360 | [TK]D-Fender: Thanks. You think the bootcamp session is overkill? |
| 18:10.30 | [TK]D-Fender | waverly360: I'm not sure of its full content VS what you need. |
| 18:12.02 | jmls | [TK]D-Fender: damn. AGI newbie. All I want is to post the queue info to a url :( |
| 18:13.34 | [TK]D-Fender | jmls: By dialing an exten? |
| 18:13.44 | [TK]D-Fender | jmls: Just make a web script that pulls it via AMI |
| 18:14.00 | [TK]D-Fender | jmls: Thats how I get live queue stats on my Polycom phones. |
| 18:14.45 | jmls | food for thought |
| 18:15.04 | *** join/#asterisk marl (n=marl@89.241.242.164) |
| 18:15.30 | jmls | I was wanting to push the info as it was updated, rather than constantly pulling it. However, that may be the only way for me |
| 18:15.51 | *** join/#asterisk rpm (n=russell@75.153.47.179) |
| 18:16.35 | marl | hi folks, anyone using the IAXy adapters? want to know if its posibly to program it without using linux, or if not, wat ports does it use for rporgramming so i can open them up on my router? (using IAXy on a windoz only network :( ) |
| 18:16.46 | [TK]D-Fender | jmls: Well it need only poll when you actually refresh the page. That means it won't be polling constantly for nothing. |
| 18:17.09 | [TK]D-Fender | marl: iaxy = bleh. |
| 18:17.45 | jmls | the url to post to is not actually a web page, but an application process. Thanks for the info and thoughts, though. |
| 18:18.12 | marl | lol, ive not had meny probs with it, its one of the only ways of connecting analog to iax thow :( |
| 18:21.04 | *** join/#asterisk JerJer (n=PhatJ@pdpc/supporter/bronze/jerjer) |
| 18:21.24 | [TK]D-Fender | jmls: Depends what level of frequency you'd need as to how I'd advise doing it. |
| 18:23.15 | waverly360 | [TK]D-Fender: Cool. I'll probably end up calling them and getting a quick run-through of what they'll be covering. I just have a couple of guys that I need to get up to speed in a hurry so that I don't have to field all of the troubleshooting issues myself. Thanks :) |
| 18:33.16 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 18:34.48 | *** part/#asterisk flujan (n=flujan@200-160-115-020.static.spo.ctbc.com.br) |
| 18:37.38 | *** join/#asterisk Ad-Hoc (n=nimbus@ppp188-98.adsl.forthnet.gr) |
| 18:38.08 | *** join/#asterisk BezNalogov (n=arjan@cust-148-3.dsl.versateladsl.be) |
| 18:38.45 | russellb | marl: you can use asterisk to provision the iaxy |
| 18:38.54 | russellb | marl: using /etc/asterisk/iaxprov.conf i think |
| 18:39.18 | marl | yup, but i need to be able to forward the network to the iaxy device |
| 18:39.40 | *** join/#asterisk _ys (n=yuri@80.70.236.69) |
| 18:39.42 | shido6 | forward the network to the iaxy device? |
| 18:39.44 | marl | aserisk is on a public server, iaxy is on my frends home network , with only windoz machines :( |
| 18:39.46 | shido6 | what do you mean by that? |
| 18:40.06 | russellb | i think someone made a windows provisioning utility |
| 18:40.12 | shido6 | you could dmz the iaxy temporarily - then provision it..... then turn off the iaxy - and take the iaxy out of the dmz then turn on the iaxy |
| 18:40.14 | marl | to profision from asterisk server would require forwarding port on lan to iaxy device |
| 18:40.15 | Qwell | russellb: that's scary |
| 18:40.24 | BezNalogov | I have installed asterisk on a server at a NOC. Because I have a very shitty firewall/router I installed openvpn on that server too, I made my device (grandstream 2000) connect through the VPN gateway, but somehow it will not connect to asterisk that way.. The address on the VPN of the asterisk server is 10.10.0.1. If I use the public ip# then it does connect. The VPN itself functions perfect, I can access any service on the server, exc |
| 18:40.28 | russellb | or someone could port gtkiaxyprov to windows :) |
| 18:40.45 | marl | have just found a windoz iaxy prof tool |
| 18:40.56 | russellb | awesome |
| 18:40.57 | marl | will see if that will work |
| 18:41.03 | shido6 | or use virtualbox on the windows system :) |
| 18:41.17 | shido6 | and provision inside your lan |
| 18:41.19 | marl | should be good if it does, would make setting up single users a LOT easier :) |
| 18:44.43 | [TK]D-Fender | BezNalogov: first suspicion : You need to add the other subnet to your localnet clause' |
| 18:45.22 | shido6 | anyone want a pri? |
| 18:45.54 | *** join/#asterisk Greek-Boy (n=email@41.221.58.2) |
| 18:45.57 | BezNalogov | <PROTECTED> |
| 18:45.58 | nestAr | sure, i could use one at my house. ;) |
| 18:49.32 | *** part/#asterisk deeperror (n=deeperro@69-215-202-195.ded.ameritech.net) |
| 18:49.41 | [TK]D-Fender | BezNalogov: First, please do you repeat yourself incessantly just because noone has answered your question in 5 minutes. Second... I DID. |
| 18:49.43 | jameswf | jbot, poop |
| 18:49.44 | jbot | ACTION fertilizes the channel |
| 18:51.44 | *** join/#asterisk CBU[^_^]M`` (n=love@210.213.142.86) |
| 18:51.48 | fetcher | will an SPA-2102 handle two G729 calls at a time, or only one? |
| 18:51.57 | fetcher | I know the PAP2-NA is limited to just one |
| 18:52.00 | *** join/#asterisk duki (n=duki@host-85-27-58-159.brutele.be) |
| 18:52.07 | [TK]D-Fender | fetcher: I think they increased the RAM and supports 2 now... |
| 18:52.10 | BezNalogov | No, I repeated because I got some message from nickserv about identifying myself on this channel, so I thought my message was perhaps not posted |
| 18:52.47 | BezNalogov | I didn't see your answer, must have read over that |
| 18:52.49 | [TK]D-Fender | BezNalogov: If you were refused access you'd have been alone in an empty version of it. |
| 18:52.57 | [TK]D-Fender | <[TK]D-Fender>BezNalogov: first suspicion : You need to add the other subnet to your localnet clause' |
| 18:54.47 | BezNalogov | localnet=10.0.0.0/255.0.0.0 |
| 18:56.30 | BezNalogov | That should be correct for the 10.10.0.1 address I think |
| 18:57.01 | [TK]D-Fender | BezNalogov: Should be. Check your firewall and what port & protocols can pass over your VPN. |
| 18:57.11 | [TK]D-Fender | BezNalogov: Then attempt some traces |
| 18:57.20 | jameswf | all my exes live in pbx's thats why I make my home.... |
| 18:57.44 | *** join/#asterisk NightMonkey (n=NightMon@pdpc/supporter/sustaining/NightMonkey) |
| 19:00.22 | *** join/#asterisk snook3r (n=snook3r@bzq-219-46-202.isdn.bezeqint.net) |
| 19:01.09 | nestAr | lol @ jameswf |
| 19:01.34 | BezNalogov | On the firewall port 1194 is forwarded to the VPN gateway |
| 19:01.40 | BezNalogov | NAT forward |
| 19:02.36 | [TK]D-Fender | BezNalogov: Make sure you have a route (static or default) what ID's the other side |
| 19:02.56 | [TK]D-Fender | BezNalogov: Shouldn't be NAT'd for a VPN |
| 19:07.19 | jameswf | i see sineapps went bye bye |
| 19:07.52 | *** join/#asterisk fernando (i=fernando@unaffiliated/musb) |
| 19:07.53 | [TK]D-Fender | jameswf: long gone & ADN is on VentureVoip |
| 19:08.10 | fernando | how to use md5 auth instead plain text in sip.conf? |
| 19:08.24 | BezNalogov | The asterisk server is connected to a public ip#, it gets the local ip# through the tun device. Is that info useful perhaps? |
| 19:08.24 | jameswf | Im always the lastt to know... |
| 19:08.44 | BezNalogov | The routing works fine, I can access any service on the server, just not asterisk |
| 19:11.51 | [TK]D-Fender | BezNalogov: Make sure the TUN is up before * starts so it binds to that IP. |
| 19:19.12 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 19:21.37 | *** join/#asterisk cfh (n=luca@195.206.30.210) |
| 19:21.45 | *** join/#asterisk izaak (n=izaak@modemcable132.248-130-66.mc.videotron.ca) |
| 19:22.08 | [TK]D-Fender | BezNalogov: Wow, positively sick pricing on Cisco's : http://www.ipphone-warehouse.com |
| 19:23.57 | [TK]D-Fender | 7961G @ 185$ USD |
| 19:23.57 | *** join/#asterisk izaak (n=izaak@modemcable132.248-130-66.mc.videotron.ca) |
| 19:23.57 | cfh | hi all , i have a problem with asterisk manager with a script perl |
| 19:23.57 | cfh | if i try with telnet it works good |
| 19:23.58 | *** join/#asterisk Darthclue (n=chatzill@zeus.nisd.net) |
| 19:23.58 | cfh | with perl and Net::Telnet it doesnt work |
| 19:27.20 | lirakis | can you forcibly unregister a sip peer via cli |
| 19:27.21 | lirakis | ? |
| 19:27.53 | De_Mon | will Set(_CDR(userfield)=something get inherited like a normal channel variable? |
| 19:28.39 | [TK]D-Fender | lirakis: not in any sane way |
| 19:28.58 | lirakis | [TK]D-Fender: okay |
| 19:30.59 | [TK]D-Fender | De_Mon: No, thats not a var, its a function. |
| 19:31.27 | [TK]D-Fender | De_Mon: depending how you recurse yuo can copy it to a real var and post it back... |
| 19:33.03 | blitzrage | Set(_MY_CDR_USERFIELD=something) |
| 19:33.58 | *** join/#asterisk w3pog (n=pgrace@aeneas.fierymoon.com) |
| 19:34.00 | blitzrage | ExecIf($[${EXISTS(${MY_CDR_USERFIELD})}]?Set(CDR(userfield)=${MY_CDR_USERFIELD}):NoOp(Nothing to do))}) |
| 19:34.10 | blitzrage | maybe something like that... :) |
| 19:34.24 | km- | that's a pretty righteous if statement |
| 19:34.26 | blitzrage | (on the other channel to see that the field was set, and if so, to Set(CDR(userfield)=...) |
| 19:34.37 | blitzrage | km-: I've created much larger expressions... |
| 19:34.56 | km- | nice. |
| 19:35.07 | km- | ok, so I've got a bit of a conundrum |
| 19:35.19 | km- | g711 is supposed to be 64kbps per call, right, 32kbps per audio direction. |
| 19:35.28 | blitzrage | I think that'll only work in trunk because the format for ExecIf() is slightly different in 1.4... so I'd have done it a slightly different way in 1.4 |
| 19:35.42 | blitzrage | km-: no -- 64kb/s per direction I believe |
| 19:35.49 | km- | no shit. |
| 19:35.54 | blitzrage | I've actually never even thought about it that way... :D |
| 19:36.19 | km- | well, every B channel on a PRI is 64kbps, isnt it? |
| 19:36.36 | km- | so one phone call in g711 should technically fit in a single B channel (if we threw away ip overhead) |
| 19:36.48 | blitzrage | hrmmm... I suppose that makes sense :) |
| 19:37.12 | km- | we need the yogi of voip to explain this one |
| 19:37.14 | km- | heh |
| 19:37.29 | km- | need some jerjer or jtodd action |
| 19:37.44 | blitzrage | 1.4: Exec(${IF($[${EXISTS(${MY_CDR_USERFIELD})}]?Set(CDR(userfield)=${MY_CDR_USERFIELD}):NoOp(Nothing to do))}) |
| 19:37.56 | blitzrage | my brain is not working today very well, heh |
| 19:38.00 | blitzrage | need a nap or something |
| 19:38.11 | km- | on a completely unrelated note, I have setup freenum and gotten an assignment, but yet to find anyone to call with it. |
| 19:38.13 | [TK]D-Fender | blitzrage: And of course it'll only work if you have that tacked onto whatever exten you're calling.. hope you aren't aiming too "wide" :p |
| 19:39.02 | blitzrage | [TK]D-Fender: I don't understand what you mean. Of course it'll only work on the channel you are calling... that's the point of an inherited channel variable... |
| 19:39.47 | [TK]D-Fender | blitzrage: I mean the ability to set it depdnds on his instering that code into every exten he MIGHT dial into thus making his diaplan look pretty cludgy potentially... |
| 19:40.34 | blitzrage | well, it'd just be part of a GoSub() or something, depending how he calls his extensions. What you just said could be true about every single suggestion. |
| 19:41.03 | [TK]D-Fender | blitzrage: I never said there would be a cleaner way :) |
| 19:41.18 | [TK]D-Fender | blitzrage: Just advertising that "life sucks" |
| 19:41.28 | blitzrage | I never said that you said there would be - just that your statement was kinda redundant |
| 19:42.02 | blitzrage | s/redundant/implied |
| 19:42.27 | [TK]D-Fender | blitzrage: Only to those who already knew the answer before you posted it :) |
| 19:42.31 | *** join/#asterisk aikanaro79 (n={aikanar@89.181.75.200) |
| 19:42.54 | [TK]D-Fender | blitzrage: Much like Jack Sparrow's compass.... only be found by someone who already knows where it is :) |
| 19:43.02 | blitzrage | well, he asked if he could create an inherited function -- I was showing how it would be done if you needed something inherited. However he implements it is not my issue :) |
| 19:43.22 | aikanaro79 | hi everyone...does anyone know if there is any irc channel for voip/sip related questions? |
| 19:43.27 | De_Mon | I watched Pirates 3 yesterday I loved that last scene where jack is tryin to find the way to.. somewhere and the compass points to his rum |
| 19:43.49 | De_Mon | aikanaro79 there is none |
| 19:44.04 | De_Mon | aikanaro79 we only talk about asterisk related questions in here, sorry |
| 19:44.30 | aikanaro79 | thanks anyway De_Mon |
| 19:44.36 | blitzrage | outtolunc: if I used ISNULL(), you'd just switch the ?<true>:<false> stuff around, but it makes more sense to use EXISTS() in that situation (at least it seems to be easier to read in my mind) |
| 19:45.24 | De_Mon | blitzrage does exists check for null? or will it return true on "" |
| 19:45.38 | blitzrage | De_Mon: it returns TRUE if it EXISTS |
| 19:46.01 | blitzrage | aikanaro79: go ahead and ask -- just keep it generic and not specific to a piece of software or hardware, and you might get away with it :) |
| 19:46.21 | lirakis | how do i show channel detail again? .. im blanking and not seeing it in help |
| 19:46.34 | De_Mon | exactly, so isnull would be "better" and by better I mean, checking that the variable actually has something meaningful in it to pass into CDR(userfield) |
| 19:46.34 | lirakis | sip channel |
| 19:46.34 | blitzrage | De_Mon: otherwise, it returns FALSE |
| 19:46.35 | km- | I wonder if anyone's considered using mp3 for voip compression in calls.. I bet it wouldn't be fast enough |
| 19:46.36 | Darthclue | lirakis, show channel |
| 19:46.40 | blitzrage | (well, it returns 1 or 0) |
| 19:46.53 | Darthclue | lirakis, sip show channel |
| 19:47.05 | De_Mon | how it would get set and then set to string null is beyond me but thats beside the point |
| 19:47.17 | De_Mon | not how so much as why |
| 19:47.22 | blitzrage | De_Mon: if you look at the code, EXISTS() and ISNULL() are just opposites of each other -- it makes more sense to me to do something if it the value exists, otherwise, do No Operation |
| 19:47.37 | De_Mon | really? so ISNULL is more of a NOTEXISTS() |
| 19:47.40 | aikanaro79 | well...I have this problem...I'm supposed to develop a client app for VoIP conference using SIP and it is supposed to interact with Asterisk...I got the dialplan right...However how can one dial a number and that number reach diferent people? |
| 19:47.41 | nestAr | love the cable company.. |
| 19:47.41 | blitzrage | De_Mon: if the variable is not set, then EXISTS() returns 0 |
| 19:47.45 | nestAr | tv just went out.. |
| 19:47.51 | lirakis | arg |
| 19:47.54 | blitzrage | Set(MY_NULL_VALUE=) |
| 19:47.58 | nestAr | guess if i had cable internet, i wouldn't be talking. |
| 19:48.03 | blitzrage | that would return TRUE with ISNULL() |
| 19:48.04 | lirakis | i keep getting xXXXXXX is not a known channel when i try to do a soft hangup |
| 19:48.10 | blitzrage | ISNULL() would also return TRUE if it wasn't set at all |
| 19:48.14 | De_Mon | what about exists |
| 19:48.30 | Darthclue | aikanaro79, you can dial multiple people using the dial command |
| 19:48.31 | *** part/#asterisk myiagy (n=myiagy@200.215.59.112) |
| 19:48.34 | De_Mon | i've never really thought about unsetting a variable, is that how you do it? |
| 19:48.43 | blitzrage | EXISTS() returns TRUE if the variable is set and not null. EXISTS() returns FALSE if the variable is not set, or set to null) |
| 19:48.46 | De_Mon | Darthclue only one person will pickup tho |
| 19:48.47 | blitzrage | De_Mon: yes |
| 19:48.53 | km- | I need to hack on lumenvox some more |
| 19:48.58 | aikanaro79 | right...but how do you code the request?...a SIP request is supposed to have only one destination |
| 19:49.10 | aikanaro79 | and I'm figuring that's what asterisk is expecting |
| 19:49.21 | blitzrage | aikanaro79: Asterisk would send multiple INVITEs out -- 1 to each destination |
| 19:49.24 | De_Mon | I submit that the function ISNULL be renamed to NOTEXISTS! |
| 19:49.35 | Darthclue | De_Mon, you're right... |
| 19:49.44 | [TK]D-Fender | aikanaro79: How would you have USER of said app do it? |
| 19:49.46 | blitzrage | ISNULL() is easier to type than NOTEXISTS() |
| 19:49.48 | *** join/#asterisk analyysi (n=ayrjola@cs181173201.pp.htv.fi) |
| 19:49.54 | blitzrage | and easier to read |
| 19:50.00 | jameswf | greedy bastards wanna be bleeding edge and stable wtf |
| 19:50.06 | De_Mon | ISN is even easier to write and even less obvious |
| 19:50.13 | aikanaro79 | TK that's my problem...that's where I'm stuck....my user sees a list of registered users |
| 19:50.17 | blitzrage | EXISTS() and ISNULL() are quite obvious |
| 19:50.30 | aikanaro79 | then...through point-and-click he chooses the ones he wants to talk to |
| 19:50.42 | [TK]D-Fender | jameswf: My "cutting edge" : http://gallery.aocomputing.net/index.php?album=2007-03-02+Oni+Forge+Bushi |
| 19:50.58 | [TK]D-Fender | aikanaro79: Then have each choose the exten you want it to lead to. |
| 19:51.17 | Darthclue | aikanaro79, is this a one to one conversation or a one to many? |
| 19:51.29 | aikanaro79 | 1-to-many |
| 19:51.33 | *** join/#asterisk c3101 (n=c3101@wbs-41-208-248-202.wbs.co.za) |
| 19:51.41 | lirakis | i basically have a sip peer that i cant seem to unregister |
| 19:51.44 | c3101 | hi ppl |
| 19:51.45 | lirakis | ive tried restarting asterisk |
| 19:51.47 | jameswf | I saw one of those get f-d up on mythbusters |
| 19:51.50 | blitzrage | in Asterisk you would have to do it with Local channels and drop the answered extensions into a MeetMe() |
| 19:51.52 | De_Mon | [TK]D-Fender does that cat live with you? |
| 19:51.56 | c3101 | need some help pls |
| 19:51.58 | analyysi | Hi, could someone help me find out where call waiting is defined? |
| 19:52.06 | aikanaro79 | blitz I use app_conference |
| 19:52.20 | [TK]D-Fender | De_Mon: Which cat? Referring to my Katana, or the other photo album? |
| 19:52.33 | blitzrage | Dial(Local/first_phone@conferenced_call&Local/second_phone@conferenced_call) |
| 19:52.34 | aikanaro79 | but I can only get 2 people in the same room...other users have to dial the room explicitly |
| 19:52.35 | De_Mon | the other photo album |
| 19:52.51 | [TK]D-Fender | jameswf: No idea exactly which makes they used, so naturally I don't trust them. |
| 19:52.58 | blitzrage | then in the [conferenced_call] you have it call the phones, then put them in a conference upon answer |
| 19:53.19 | blitzrage | I think that'd work -- I could be very wrong :) My brain is off today. |
| 19:53.25 | c3101 | trying to have * talk to an avaya (ugh) with sip, the avaya only seems to support tcp/tls, and * does quite the opposite, has anyone mannaged to get * to talk to avaya ? |
| 19:53.26 | De_Mon | whoa lotsa albums |
| 19:53.28 | [TK]D-Fender | blitzrage: EW. |
| 19:53.37 | blitzrage | Better way would be to write an application to do it via the AMI |
| 19:53.44 | aikanaro79 | but blitz that's dialplan logic...that's not the problem for now...(unless I'm mistaken)...how do I reach asterisk with the request? |
| 19:53.54 | blitzrage | with an INVITE.... ? |
| 19:53.58 | [TK]D-Fender | De_Mon: Ah, those are my sister's Servals. African Wildcats. Don't have them any more.... |
| 19:53.58 | aikanaro79 | exactly |
| 19:54.09 | blitzrage | you send multiple INVITEs -- 1 for each end point |
| 19:54.28 | blitzrage | you don't send 1 INVITE for multiple end points |
| 19:54.45 | aikanaro79 | ok...and how does asterisk know it's all the same conference and not diffente conferences? |
| 19:55.01 | De_Mon | [TK]D-Fender did the eat a neighors dog or something ;) |
| 19:55.06 | *** join/#asterisk beasty (n=beasty@about/apple/macbook/beasty) |
| 19:55.08 | blitzrage | that would be dependent upon the dialplan logic |
| 19:55.29 | beasty | is it possible to run a command when some one is calling ? |
| 19:55.32 | aikanaro79 | I could I mark a call so that later, inside the dialplan, I could use that idea? |
| 19:55.47 | De_Mon | c3101 there is an experimental tcp/tls branch. Another option is to use a proxy such as openser to do the tls/tcp<->udp switch |
| 19:55.53 | [TK]D-Fender | De_Mon: No, they were bought to breed and leave it to my sister to find the only 2 gay ones.... no interest in each other and as they "marked" their territory they wrecked their furniture and flooring. |
| 19:56.02 | blitzrage | aikanaro79: I don't understand what you mean -- you could mark the call with a channel variable or something I suppose |
| 19:56.09 | De_Mon | awww dang |
| 19:56.30 | aikanaro79 | blitz...asterisk could get that variable and know it's the same conference right? |
| 19:56.38 | aikanaro79 | I mean inside the dialplan |
| 19:56.40 | blitzrage | aikanaro79: depending how you program your dialplan -- yes. |
| 19:56.47 | *** part/#asterisk drmessano (n=nonya@c-76-125-26-150.hsd1.ga.comcast.net) |
| 19:56.48 | [TK]D-Fender | De_Mon: She fed them raw drumsticks.... was really creepy when you heard the bones shatter.... they ate them WHOLE. |
| 19:57.02 | aikanaro79 | is there any variable for this? I mean recommended for such uses |
| 19:57.12 | blitzrage | no -- you have to program the logic yourself |
| 19:57.18 | c3101 | De_Mon, this must go into serious production environment, the svn dev branch isn 't an option..... no way to get it going without a proxy ? maybe somebody in here knows how to make avaya talk over plain udp ?? |
| 19:57.19 | Greek-Boy | i did not find anything about chanspy in UPGRADE.txt in 1.4 so I'm assuming that it still works the same... |
| 19:57.21 | De_Mon | [TK]D-Fender I've got a golden retreiver, she eats whole chickens, raw -- bones and all |
| 19:57.47 | aikanaro79 | blitzrage, I meant channel variables...any? |
| 19:57.48 | [TK]D-Fender | De_Mon: Good = cat. Better = Dog. Best = Dog that eats cats :D |
| 19:57.56 | blitzrage | aikanaro79: you set channel variables with Set() |
| 19:58.00 | De_Mon | turkey legs are a bit more of a clannenge |
| 19:58.04 | aikanaro79 | I see |
| 19:58.12 | blitzrage | aikanaro79: you have to build the logic so Asterisk knows what you want to do with the channel |
| 19:58.17 | De_Mon | [TK]D-Fender agreed! |
| 19:58.31 | blitzrage | you might even have the information living in a database external of Asterisk, and you could pull it from func_odbc... it just depends what you're trying to accomplish |
| 19:58.37 | De_Mon | wtf s/clannenge/challenge |
| 19:58.47 | *** join/#asterisk __freedom__lover (n=eduardo@clipper.provale.com.br) |
| 19:58.49 | rob0 | Stonehenge? |
| 19:58.57 | blitzrage | c3101: Asterisk won't support TCP/TLS without patches from the bug tracker |
| 19:59.11 | aikanaro79 | blitzrage, so I send multiple invites but everyone has a variable that tags it as part of a conference...then asterisk has dialplan logic to connect all the invites and get a room for the conference |
| 19:59.15 | De_Mon | I've got an extension that when dialed creates multiple call files that dial conference members and joins them to a conference. Easy peasy |
| 19:59.17 | blitzrage | c3101: if the avaya will *only* let you use those -- then you're out of luck. Asterisk only supports UDP. |
| 19:59.23 | aikanaro79 | is this kind of what you're telling me? |
| 19:59.54 | *** join/#asterisk [N00B] (n=ckwall@206.71.78.172) |
| 20:00.17 | c3101 | this suck ! have a great opportunity to roll out 300 * pbx's, but must integrate into avaya |
| 20:00.31 | blitzrage | aikanaro79: ummm... kinda -- but the INVITE has nothing to do with what conference room to connect with unless you use something like SIPAddHeader, which adds a header and value you could parse on |
| 20:01.23 | [TK]D-Fender | aikanaro79: make those "buttons" you were referring to each dial a different extensions. 1 for the conference simply leads to the conference. |
| 20:01.24 | _x86_ | c3101: that's easy though... |
| 20:01.37 | [N00B] | i have a remote server connecting to my main asterisk server via SIP. I recently changed it from IAX2 to SIP. Now I am getting periodic errors like this one: http://pastebin.ca/784640 anyone know how that is caused or where to even begin my troubleshooting? |
| 20:01.40 | blitzrage | c3101: http://bugs.digium.com/view.php?id=4903 and http://bugs.digium.com/view.php?id=5413 |
| 20:01.41 | c3101 | i'm listening _x86_ |
| 20:01.53 | aikanaro79 | [TK]D-Fender, sorry...didn't quite get your idea |
| 20:01.56 | _x86_ | c3101: most avaya systems support SIP, or can with an IP card... which makes it very easy to trunk between the two systems |
| 20:02.18 | _x86_ | c3101: if you have an older avaya that can not support a SIP / IP card, you can trunk via T1 interfaces |
| 20:02.32 | [TK]D-Fender | aikanaro79: You said your client will have buttons for individuals and for a conference, no? |
| 20:03.06 | aikanaro79 | [TK]D-Fender, no...it only has to support conference calls |
| 20:03.09 | [TK]D-Fender | [N00B]: First guess... * is behind NAT and wasn't set up properly for it. |
| 20:03.13 | c3101 | so far only been able to make avaya talk sip over tcp/tls _x86_ , if you know howto make it talk plain udp, i'd give you a case of beer |
| 20:03.26 | aikanaro79 | stupid I know...but those are not my rules |
| 20:03.31 | [TK]D-Fender | aikanaro79: You mean as a client has to be able to bring someone into an existing call? |
| 20:03.33 | [N00B] | [TK]D-Fender: ok. i will take a look there. |
| 20:03.38 | _x86_ | c3101: call avaya and ask them how to disable sRTP |
| 20:04.12 | aikanaro79 | [TK]D-Fender, you're a client that need to talk to 4 different people...you choose them from the user list and get into a conference call with them |
| 20:04.17 | *** join/#asterisk Victor_Yure (n=aaa@postfix.tradein.com.br) |
| 20:04.22 | aikanaro79 | this is how it's supposed to work |
| 20:04.40 | c3101 | that's like asking your gran'ma to change the alternator on your car m8 ! |
| 20:04.50 | [TK]D-Fender | aikanaro79: 3-way (2 others) = possible. Anything else = comlpicated. |
| 20:05.01 | [N00B] | [TK]D-Fender: no nat. all on private network |
| 20:05.08 | [N00B] | i have nat=no on my sip entry |
| 20:05.14 | Darthclue | aikanaro79, use call files. Should be easy to do. |
| 20:05.23 | aikanaro79 | [TK]D-Fender, unfortunately I got a similar question from asterisk developers...nevertheless it should be possible |
| 20:05.32 | aikanaro79 | Darthclue, sorry...what are those? |
| 20:05.42 | aikanaro79 | blitzrage, yes...I think that was what I was talking about...evengthough all INVITES have different destinations, dialplan logic calculates a conference room right? |
| 20:06.11 | blitzrage | aikanaro79: ya, I guess so |
| 20:06.24 | [TK]D-Fender | aikanaro79: unless your client will force redirects of all the appropriate people to that room I don't see a way |
| 20:07.06 | [TK]D-Fender | [N00B]: check your firewall settings, etc then. |
| 20:07.09 | *** join/#asterisk aikanaro79 (n={aikanar@89.181.75.200) |
| 20:07.25 | krondorl | Question: i would like to add a c program in the apps directory of 1.2.21 of asterisk, but when I compile it, the code does not get picked up. Does it need to be defined somwhere in the makefile for it to be picked up? |
| 20:07.47 | Greek-Boy | I have a WIP330 phone on a wireless network behind a firewall. If I set it to register to asterisk do I still need firewall rules? outbound connections are all unfiltered on the firewall. |
| 20:07.48 | [TK]D-Fender | krondorl: Yes |
| 20:07.51 | Darthclue | aikanaro79, take a look at the book, Chapter 14 |
| 20:07.51 | aikanaro79 | [TK]D-Fender, not even with blitzrage's suggestion? |
| 20:07.53 | Darthclue | ~book |
| 20:07.53 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf |
| 20:08.12 | [TK]D-Fender | aikanaro79: Which was? |
| 20:08.15 | krondorl | [TK]D-Fender: :) ok where? cus I cannot find it.. |
| 20:08.20 | blitzrage | aikanaro79: see the G() option of the Dial() application |
| 20:08.25 | De_Mon | aikanaro79 you just want to dial an extension lets say 500, and then have asterisk call a predefined set of devices and join them all to a random conference number |
| 20:08.27 | [TK]D-Fender | krondorl: MakeFile in your apps folder. |
| 20:08.42 | krondorl | DOH.. I only looked in the main folder.. |
| 20:08.44 | *** part/#asterisk cfh (n=luca@195.206.30.210) |
| 20:08.51 | [TK]D-Fender | krondorl: Same way you patch for SpanDSP's rxfax/txfax |
| 20:08.56 | aikanaro79 | [TK]D-Fender, mark every INVITE of the same conference with a variable and later dialplan logic redirects every user to the same room |
| 20:09.26 | krondorl | [TK]D-Fender: :) Ummm, ok... Sorry this is the first time I have ever done this... |
| 20:09.36 | krondorl | never needed to before.. |
| 20:09.50 | blitzrage | Dial(SIP/some_phone,30,G(conference_calls,s,1)) might work |
| 20:09.58 | blitzrage | [conference_calls] |
| 20:09.59 | aikanaro79 | De_Mon, not exactly...I want to dial several users and have them all in the same conference room....dialplan can't possibly know in advance which users are going to be called |
| 20:10.02 | c3101 | damn, okay, tnx anyway |
| 20:10.18 | [TK]D-Fender | aikanaro79: If your app is doing the invite, then IT will have to do a redirect to the conference. |
| 20:10.32 | blitzrage | exten => s,1,MeetMe(${SIP_HEADER(X-conference_room)}) |
| 20:10.43 | blitzrage | exten => s,2,MeetMe(${SIP_HEADER(X-conference_room)}) |
| 20:10.47 | [TK]D-Fender | aikanaro79: Other option is to initiate a "call file" as Darthclue suggested that upon them annswering will throw them into a conference as it is |
| 20:11.12 | blitzrage | aikanaro79: I think the above will work assuming your INVITEs contain the X-conference_room header |
| 20:11.17 | aikanaro79 | [TK]D-Fender, I have to look up call files |
| 20:11.36 | blitzrage | then you can use the above suggestion as I just outlined |
| 20:11.38 | aikanaro79 | blitzrage, i'd prefer app_conference as it does not need an external timer |
| 20:12.04 | aikanaro79 | but still...I have to look up that G() option |
| 20:12.04 | blitzrage | whatever! Replace MeetMe() with whatever application you're using. |
| 20:12.19 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 20:12.20 | *** join/#asterisk geekrebel (n=CmaX@dsl-241-37-57.telkomadsl.co.za) |
| 20:12.22 | blitzrage | MeetMe() could be SomeWeirdForeignApplicationFromSweden() |
| 20:12.31 | blitzrage | you're looking too much into the specifics |
| 20:12.48 | blitzrage | as opposed to the overall logic |
| 20:12.50 | geekrebel | ahoy folks! |
| 20:12.55 | blitzrage | howdy |
| 20:12.55 | aikanaro79 | blitzrage, sorry...this is the only help I'm getting on this |
| 20:13.02 | geekrebel | blitzrage: :) |
| 20:13.11 | De_Mon | geekrebel hoi |
| 20:13.34 | geekrebel | quick question: (before I ask AsteriskNow specific questions) Is the Web interface for AsteriskNow and Asterisk the same? (more or less) |
| 20:13.35 | *** join/#asterisk ZenBSDi (n=zenbsdi@unaffiliated/ZenBSDi) |
| 20:13.42 | geekrebel | De_Mon: ahoy! |
| 20:13.43 | blitzrage | aikanaro79: I think what I outlined above will do what you expect assuming you can pass information into Asterisk somehow at all setup time -- like with an X header which you retrieve with the SIP_HEADER() function |
| 20:14.02 | blitzrage | s/all/call/ |
| 20:14.13 | blitzrage | gah -- I should have left the / off the end |
| 20:14.20 | [TK]D-Fender | geekrebel: It is only for AsteriskNOW. |
| 20:14.31 | [TK]D-Fender | geekrebel: Or rather the GUI. |
| 20:14.38 | De_Mon | I'd be less inclined to use sip-headers and more inclied toward channel variables |
| 20:14.41 | [TK]D-Fender | geekrebel: And yes it is verboten here... |
| 20:14.57 | blitzrage | De_Mon: where do you set the channel variables though if the call is coming from an external resource |
| 20:14.58 | [TK]D-Fender | De_Mon: I'd be more inclined towards EXTENS <- |
| 20:15.03 | geekrebel | [TK]D-Fender: dang! ;-) |
| 20:15.04 | aikanaro79 | why is it necessary a different context? G option redirects a call right? |
| 20:15.10 | blitzrage | aikanaro79: yes |
| 20:15.20 | geekrebel | [TK]D-Fender: the asterisknow channel is so small! |
| 20:15.30 | blitzrage | ok -- I give up on this issue. I think everyone has given more than enough suggestions of how to approach it |
| 20:15.52 | [TK]D-Fender | geekrebel: Wanna use a chump GUI and don't like the support? Cry me a river.... so I can hold your head under.... |
| 20:15.55 | De_Mon | aikanaro79 G() redirects caller and calling party into a different context,exten,priority... |
| 20:16.08 | *** join/#asterisk atomicd (n=Atomicd@74-206-0-80.static-ip.m.telepacific.net) |
| 20:16.15 | aikanaro79 | De_Mon, I'm thinking...that's why I'm not answering |
| 20:16.28 | aikanaro79 | but thanks nevertheless |
| 20:16.29 | aikanaro79 | :) |
| 20:16.40 | russellb | [TK]D-Fender: be nice. |
| 20:17.25 | russellb | plz :) |
| 20:17.30 | De_Mon | aikanaro79 you could just as easily dial an extension like _XXX where XXX is a conference *you pick* and sends whoever you called there using a call file, theres at least 5 distinct ways to acomplish this just pick the one you like best :) |
| 20:18.26 | aikanaro79 | De_Mon, I have to understand them so I can choose one :) (I'm still quite a newbie regarding asterisk) |
| 20:19.22 | aikanaro79 | blitzrage, I can't check my dialplan right now but I think it's similar to your logic...I lacked the variable idea |
| 20:20.24 | JayTee52 | I'm getting the error: "Got SUBSCRIBE for extensions without hint. Please add hint to XXXX in context default." Can anyone explain this one to me? |
| 20:20.39 | aikanaro79 | thanks everyone for your input...I'll just dive right in |
| 20:21.25 | [TK]D-Fender | JayTee52: A device you reg'd to * is trying to get the "on the phone" status of an exten that does not have a HINT (watcher) set up |
| 20:27.33 | De_Mon | aaie |
| 20:28.01 | De_Mon | [Nov 19 15:30:28] NOTICE[6523]: app_meetme.c:1912 conf_run: Audio bytes: 80 Buffer size: 320 |
| 20:28.04 | De_Mon | [Nov 19 15:30:30] NOTICE[6523]: app_meetme.c:1912 conf_run: Audio bytes: 320 Buffer size: 80 |
| 20:28.36 | De_Mon | meetme somehow got mixed up and is off-by-one on the buffer size/audio bytes |
| 20:28.46 | De_Mon | poor conference |
| 20:29.41 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:31.42 | jameswf | f |
| 20:34.47 | *** join/#asterisk xheliox (n=jeff@pdpc/supporter/active/xheliox) |
| 20:39.51 | jameswf | anyone use evolution for email |
| 20:40.49 | geekrebel | used to |
| 20:41.02 | geekrebel | few years ago |
| 20:41.12 | jameswf | tbird is poopin |
| 20:41.17 | *** join/#asterisk r0d3nt (i=nobody@foster.stonedcoder.org) |
| 20:41.32 | geekrebel | roelf! |
| 20:41.50 | De_Mon | replace the newspaper |
| 20:42.21 | [TK]D-Fender | jameswf: Tbird not doing so good? They jsut put out 2.0.0.9 |
| 20:42.56 | [TK]D-Fender | Is there a decent free build of Evolution for Win32 yet? |
| 20:43.03 | [TK]D-Fender | been a while since I looked. |
| 20:43.28 | jameswf | dont use win32 |
| 20:43.57 | jameswf | *I |
| 20:44.13 | [TK]D-Fender | jameswf: Wasn't targeted at you :) |
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| 20:47.32 | *** join/#asterisk etfonhomey (n=chatzill@mail.advancmed.org) |
| 20:47.39 | *** join/#asterisk pepo-- (n=pepOSX@190.78.220.149) |
| 20:48.43 | [TK]D-Fender | Google has seved me well :) |
| 20:48.51 | [TK]D-Fender | served* |
| 20:48.52 | krondorl | If I have a Wait() before a Playback() I should wait before anything is played right?? |
| 20:49.17 | [TK]D-Fender | krondorl: Yes. |
| 20:50.17 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 20:51.01 | *** join/#asterisk riddlebox (n=james@75-128-170-26.static.stls.mo.charter.com) |
| 20:53.19 | De_Mon | what does this mean Local/700@parkedcalls-30de,1<ZOMBIE> |
| 20:53.44 | De_Mon | krondorl It waits for x seconds such as Wait(x) |
| 20:53.55 | De_Mon | I dont think Wait() will actually do anything |
| 20:54.15 | De_Mon | you can also use Wait(.5) |
| 20:54.45 | krondorl | De_Mon: lol, I know how long it waits I was more worried that the playback ignores the wait.. I sometime get mixed up with background and playback on how they react.. |
| 20:55.15 | De_Mon | priority doesnt leave wait till the time has elapsed |
| 20:55.20 | krondorl | I left the insides of () on purpose.. |
| 20:55.35 | krondorl | That's good to know.. |
| 20:56.55 | *** join/#asterisk jtknapp (n=skip@65-126-63-1.dia.static.qwest.net) |
| 20:56.56 | De_Mon | krondorl you could also do a Playback(silence/1&yourfile) I do believe. |
| 20:57.31 | *** join/#asterisk mackes (n=root@65-121-253-83.dia.static.qwest.net) |
| 20:57.57 | krondorl | interesting idea... Client is testing a wait(4) right now to see if that's enough.. |
| 21:07.12 | De_Mon | krondorl what are they waiting for? |
| 21:07.43 | krondorl | AMD before the message is played back to the AM.. |
| 21:08.00 | De_Mon | ooh |
| 21:08.26 | krondorl | part of the beginning of the message is being cut off. |
| 21:08.37 | [TK]D-Fender | krondorl: at which point you want to make sure the channel is answered. For that you'd probably want to use Playback(silence/4) |
| 21:08.44 | *** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177584588.dsl.bell.ca) |
| 21:08.52 | [TK]D-Fender | krondorl: to ensure the audio path is up. |
| 21:09.04 | De_Mon | krondorl you could also do a Playback(silence/4&yourfile) I do believe. !!!! |
| 21:09.06 | [TK]D-Fender | krondorl: Wait != answer. |
| 21:09.08 | *** join/#asterisk ghento (n=ghento@75.155.241.7) |
| 21:09.12 | [TK]D-Fender | De_Mon: Yes |
| 21:09.27 | krondorl | channel is already active.. it's part of vicidial... |
| 21:09.28 | De_Mon | I sense you are not a fan of multiple files in one playback command |
| 21:09.33 | [TK]D-Fender | De_Mon: eash has their advantages |
| 21:09.52 | De_Mon | [TK]D-Fender any advantages besides astetics? |
| 21:10.53 | De_Mon | aestitic ones |
| 21:10.53 | [TK]D-Fender | De_Mon: being split reads a little easier as its sequential visually. easier to grpe for changes, etc. |
| 21:11.01 | [TK]D-Fender | De_Mon: Very rare for a truly solid case one way or the other. |
| 21:11.27 | De_Mon | thats because the reasons are mostly emotinal or personal taste reasons |
| 21:11.47 | De_Mon | rawr my keyboard is purposefully misspelling words on me its posessed! |
| 21:12.41 | *** join/#asterisk asdx (n=diego@adsl-148-71.click.com.py) |
| 21:12.42 | asdx | hi |
| 21:12.51 | asdx | i'm getting this: -- Channel 'IAX2/teliax-9' unable to transfer |
| 21:12.51 | De_Mon | hes back quick everyone hide |
| 21:13.34 | De_Mon | my asterisk dialplan creates zombies pheer me |
| 21:13.43 | De_Mon | zombie channels that is |
| 21:14.41 | asdx | odd |
| 21:14.50 | asdx | i can call to my pstn line... |
| 21:14.56 | asdx | perfectly |
| 21:16.08 | [TK]D-Fender | asdx: enable iax2 debug and pastebin another complete call and transfer attempt |
| 21:16.18 | [TK]D-Fender | asdx: and enough core debug to see DTMF as well. |
| 21:16.44 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 21:17.18 | asdx | [TK]D-Fender: ok |
| 21:17.41 | *** join/#asterisk ajohnson (n=ajohnson@63.147.46.186) |
| 21:20.03 | marl | hi, im having a problem with my IAXy adapter, i can dial an internal 4 digit extension with no problems, but when i try and dial an outside line (11 digits) and the only thing * apears to register is the first 2 digits of the dialed number :( iax debug, for both 4 digit and 11 digit numbers and my iax.conf file, anyone any pointers? |
| 21:22.28 | *** join/#asterisk galeras (n=Martin@201.244.247.149) |
| 21:23.58 | *** join/#asterisk fnordus (n=dnall@24.84.160.227) |
| 21:24.06 | galeras | Dear Sirs, Someone knows if is possible to integrate call manager 4.1with asterisk trough h323? |
| 21:24.14 | marl | when dialing the 11 digit number, it only apears to accet the first 2 digits :( but when dialing the 4 digit number it accepts all 4 digits |
| 21:24.40 | asdx | [TK]D-Fender: http://pastebin.com/mc5d0080 |
| 21:26.27 | [TK]D-Fender | asdx: You are trying to transfer the call on your softphone? |
| 21:26.59 | asdx | [TK]D-Fender: no, i'm trying to call to spain |
| 21:27.28 | [TK]D-Fender | asdx: what is this "transfer" that is happening then? |
| 21:27.36 | *** join/#asterisk k0sm|k0 (n=k0smik0@2001:1418:1f9:babe:e:c01d:c0ca:c01a) |
| 21:28.10 | asdx | i'm not aware of that |
| 21:28.43 | k0sm|k0 | hi |
| 21:28.49 | k0sm|k0 | anyone italian heree ? |
| 21:28.52 | Greek-Boy | what "dialplan function" compensates for the loss of ${CALLERIDNUM} in 1.4? |
| 21:29.00 | [TK]D-Fender | asdx>i'm getting this: -- Channel 'IAX2/teliax-9' unable to transfer |
| 21:29.13 | [TK]D-Fender | Greek-Boy: "core show function CALLERID" |
| 21:29.21 | asdx | [TK]D-Fender: yes thats what the log says |
| 21:29.29 | ajohnson | hehe |
| 21:29.31 | [TK]D-Fender | asdx: looks like the call starts and is transferred mid-way |
| 21:29.46 | asdx | yep |
| 21:30.07 | asdx | this is what i have in my dialplan: [default] exten => 999,1,Dial(IAX2/puli2007@teliax/01134649840773) |
| 21:30.27 | *** join/#asterisk PepOSX (n=pepOSX@190.78.220.149) |
| 21:30.51 | [TK]D-Fender | asdx: Ok, I don't know if thats formatted right and the clock has run out for me here. Keep that pastebin around and maybe somebody else can pick up the ball on this one. |
| 21:31.10 | [TK]D-Fender | Ok, gtg, back later all.... |
| 21:32.06 | Greek-Boy | [TK]D-Fender: so now how would I use these functions to put data into a file name I want to generate with mixmonitor. In 1.2 I used to use variables ${CALLERIDNUM} and ${TIMESTAMP} directly on the filename... Do I now have to set each variable first? |
| 21:34.11 | De_Mon | is there a 19 character limit on accountcode or is it just my imagination? |
| 21:35.37 | De_Mon | Greek-Boy ${CALLERIDNUM} is now ${CALLERID(num)} |
| 21:40.52 | De_Mon | I grok not the new CDR mess I have created. Fudge! |
| 21:41.16 | sigmounte | anyone have experience with ztdummy and 2.6 kernel ? |
| 21:42.06 | *** join/#asterisk mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
| 21:42.06 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:43.25 | *** join/#asterisk [Outcast] (n=outcast@203-114-166-26.eth.sta.inspire.net.nz) |
| 21:45.00 | [Outcast] | has anyone every seen the queue module just deside to die with out any sort for warning? |
| 21:49.29 | De_Mon | not in 1.4 |
| 21:51.14 | [Outcast] | what about 1.2 |
| 21:52.05 | De_Mon | didnt use em much in 1.2 |
| 21:52.54 | *** part/#asterisk [N00B] (n=ckwall@206.71.78.172) |
| 21:53.13 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 21:53.52 | nestAr | been using queues since 1.0, never really had a problem with them crashing. |
| 21:54.40 | [Outcast] | you using call back agents? |
| 21:55.38 | blitzrage | [Outcast]: what version of Asterisk? |
| 21:55.55 | De_Mon | outtolunc agents? I've had all sorts of problems with agents in 1.2 |
| 21:55.56 | *** join/#asterisk bl4q (i=me@1.1.1.vg) |
| 21:56.18 | *** part/#asterisk lirakis (n=lirakis@65.200.191.253) |
| 21:56.20 | De_Mon | well, not all sorts, mostly the crashing kind :) |
| 21:56.20 | *** join/#asterisk blq (n=Bl@dslb-088-066-251-220.pools.arcor-ip.net) |
| 21:56.21 | blitzrage | there was a bug fixed in 1.4.14 with app_queue causing asterisk to hang (but not crash) |
| 21:56.44 | *** part/#asterisk blq (n=Bl@dslb-088-066-251-220.pools.arcor-ip.net) |
| 21:57.09 | De_Mon | 1.4.14 is out? how did I miss that annoucement! |
| 21:57.14 | *** join/#asterisk cfh (n=luca@195.206.30.210) |
| 21:57.19 | blitzrage | I don't know... you're a nub? :) |
| 21:57.29 | De_Mon | that hurts man |
| 21:57.40 | *** part/#asterisk cfh (n=luca@195.206.30.210) |
| 21:57.42 | De_Mon | <scout voice from tf2> |
| 21:57.55 | De_Mon | I need some asterisk tf2 players to beat up on |
| 21:58.05 | blitzrage | tf2? |
| 21:58.24 | JT | team fortress 2 i assume |
| 21:58.31 | De_Mon | http://www.youtube.com/watch?v=i68cEsALWt0 |
| 21:58.36 | blitzrage | weird... that doesn't help me any :) |
| 21:58.37 | De_Mon | yes |
| 21:58.53 | [Outcast] | 1.2.24 |
| 21:59.12 | De_Mon | link |
| 21:59.18 | blitzrage | weird. I haven't used 1.2 in over a year |
| 21:59.23 | blitzrage | sorry |
| 22:00.02 | blitzrage | I bet the bug fixed in 1.4 is in 1.2, but wasn't backported because 1.2 is in security maintenance mode only |
| 22:00.18 | blitzrage | but that's only a guess (that it is the same bug as was fixed in 1.4) |
| 22:00.47 | De_Mon | point being, its not queues but agents that are the problem |
| 22:01.17 | *** join/#asterisk [hC] (n=hardcore@66.119.167.162) |
| 22:02.30 | *** join/#asterisk ariel_ (n=ariel_@74.8.35.6) |
| 22:02.44 | ariel_ | hello everyone |
| 22:03.08 | De_Mon | http://pastebin.ca/784877 |
| 22:03.22 | *** join/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 22:04.01 | De_Mon | See those CDR records? The total call time was 60 seconds. There was a hold time of 46 seconds. |
| 22:04.14 | krondorl | Night all.. |
| 22:04.29 | De_Mon | how the hell am I supposed to determine the amount of time there was an actual agent on the call! |
| 22:04.58 | De_Mon | for each call! |
| 22:05.01 | krondorl | stick a timer up his butt?? |
| 22:05.57 | De_Mon | normally i would expect 46+31=60 but it doesnt |
| 22:06.28 | De_Mon | and 13+12+3 doesn't = 60-31 either. so none of these numbers add up |
| 22:06.37 | krondorl | huh, isn't that 77?? |
| 22:07.11 | De_Mon | actually 13+12+3 does come pretty close to 28, but close doesn't count |
| 22:07.31 | krondorl | Oh well, time to go home.. bye.. |
| 22:09.08 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 22:09.55 | De_Mon | 46-60=total time with agent |
| 22:10.57 | De_Mon | oh oh, See as soon as I talk it out with my imaginary friends on IRC it all starts to make sense |
| 22:11.08 | *** join/#asterisk cfh (n=luca@195.206.30.210) |
| 22:11.17 | *** part/#asterisk cfh (n=luca@195.206.30.210) |
| 22:11.28 | *** part/#asterisk jmls (n=jmls@62.49.235.130) |
| 22:15.43 | *** join/#asterisk Arno[Slack] (n=hellSOUN@gre92-1-81-57-177-108.fbx.proxad.net) |
| 22:16.33 | *** join/#asterisk pepo-- (n=pepOSX@190.72.148.91) |
| 22:16.50 | *** join/#asterisk cfh (n=luca@195.206.30.210) |
| 22:19.21 | *** join/#asterisk GreggB (n=GreggB@66.206.86.107) |
| 22:20.58 | cfh | hi, i have a problem with manager . If i try with telnet and " Action: Originate " i can generate a successful outcall , if i use a perl script with Net::Telenet cpan module and i send the same paramters like the session telnet i get this error : "Unable to request channel" |
| 22:21.08 | waverly360 | Hey guys...is it possible to restart a single zap device without doing a total asterisk restart? |
| 22:21.10 | *** join/#asterisk BiG^DoG (n=BiG^DoG@c-67-162-233-20.hsd1.de.comcast.net) |
| 22:21.17 | waverly360 | I have a pri, and an analog card |
| 22:21.19 | cfh | what can i do ? |
| 22:21.26 | waverly360 | I just want to change zapata info for the...well.. |
| 22:21.27 | waverly360 | nevermind |
| 22:21.31 | waverly360 | guess that's not gonna work |
| 22:22.56 | *** part/#asterisk ccaron (n=cedric@85-218-16-115.dclient.lsne.ch) |
| 22:23.51 | ariel_ | sometimes if you do from the prompt ztcfg -vvvv it will take some minor updates like caller ID info on the conf files. |
| 22:23.55 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 22:25.16 | De_Mon | waverly360 module unload/reload chan_zaptel.so will do it without restarting * |
| 22:25.26 | De_Mon | err reload or unload/load |
| 22:25.37 | waverly360 | but will it kill calls in progress? |
| 22:25.38 | *** join/#asterisk _theHub (n=_theHub@firewall.cierant.com) |
| 22:25.46 | De_Mon | only the ones over zaptel :) |
| 22:25.52 | ariel_ | yes it will kill calls on zap |
| 22:26.04 | waverly360 | that's the problem :)..but thanks |
| 22:26.15 | ariel_ | what changes did you do? |
| 22:26.41 | De_Mon | i thought so, but didn't want to guess. |
| 22:27.24 | *** part/#asterisk galeras (n=Martin@201.244.247.149) |
| 22:28.53 | blitzrage | if you change signalling, then you definitely need to restart asterisk |
| 22:29.03 | blitzrage | just an FYI |
| 22:29.34 | ariel_ | blitzrage, how are you doing? Long time, for me being here. |
| 22:29.42 | blitzrage | oh not too shabby I suppose :) |
| 22:29.49 | blitzrage | ya, I haven't seen you online in quite some time |
| 22:30.08 | ariel_ | been busy with a job I took 1.5 years ago |
| 22:30.20 | De_Mon | why doesn't signaling update with a module reload? |
| 22:30.26 | ariel_ | large 5 location call center all on asterisks systems |
| 22:30.28 | *** join/#asterisk craigk (n=ckowald@58.174.122.198) |
| 22:30.51 | ariel_ | signalling is setup on zaptel service load |
| 22:31.02 | ariel_ | which is done before loading asterisk |
| 22:31.20 | De_Mon | kernel module level eh |
| 22:32.03 | Greek-Boy | damn |
| 22:32.06 | Greek-Boy | TK is not around |
| 22:32.08 | ariel_ | blitzrage, have any ideas on mass recording of using Asterisk as Gateways with each one havign over 350 channels of sip calls? |
| 22:32.13 | Greek-Boy | anyhone know if this will work in a macro? |
| 22:32.15 | Greek-Boy | exten = > s,n,MixMonitor(internal-${CALLERID(number)}to${EXTEN}-${STRFTIME(${EPOCH}%Y%m%d-%H%M%S)}) |
| 22:33.16 | *** join/#asterisk ManxPower (n=manxpowe@69-2-85-41.wan.networktel.net) |
| 22:33.24 | Greek-Boy | basically i want to record a wave file with the filename containing the value of those variables |
| 22:35.55 | ManxPower | exten => 666,1,Record(/the/path/you/want/recording-${MYVARIABLE}.wav) |
| 22:36.00 | JT | De_Mon: you need to restart asterisk for changes in zapata singalling |
| 22:37.11 | ariel_ | hello ManxPower, long time since I been around. Hope all is going well. |
| 22:37.51 | *** join/#asterisk hfb (n=hfb@pool-71-106-219-180.lsanca.dsl-w.verizon.net) |
| 22:39.18 | *** join/#asterisk moy (n=moyhu@189.169.50.222) |
| 22:42.20 | *** join/#asterisk marc7 (n=marc@64.46.14.64.novuscom.net) |
| 22:42.35 | blitzrage | ariel_: that sounds like mostly a problem with the HD write speed -- it might help to run Asterisk in a ramdrive and sending the data over the network to another drive... ? |
| 22:42.50 | De_Mon | Greek-Boy ${CALLERID(number)} is not correct |
| 22:42.54 | blitzrage | De_Mon: yes it is |
| 22:43.08 | blitzrage | ${CALLERID(num)} and ${CALLERID(num)} are both correct |
| 22:43.09 | De_Mon | oh? /me looks again |
| 22:43.15 | blitzrage | num/number |
| 22:43.25 | Qwell | does nam work? |
| 22:43.35 | blitzrage | don't think so :) |
| 22:44.36 | De_Mon | The allowable datatypes |
| 22:44.36 | De_Mon | are "all", "name", "num", "ANI", "DNID", "RDNIS". |
| 22:44.46 | De_Mon | show function CALLERID lies! |
| 22:44.49 | blitzrage | De_Mon: if you actually look at the code, num and number both work |
| 22:45.06 | De_Mon | that, or you are... i trust you more than this outdated documentation :) |
| 22:45.28 | blitzrage | it's not outdated... it'd just be confusing to list both 'number' and 'num' in the docs |
| 22:45.47 | blitzrage | the code is just forgiving to non-standard usage |
| 22:45.49 | De_Mon | why are they both allowed besides |
| 22:45.56 | De_Mon | then its still wrong to use number |
| 22:45.57 | De_Mon | :) |
| 22:46.00 | blitzrage | no it's not |
| 22:46.09 | De_Mon | non-standard == WRONG |
| 22:46.10 | blitzrage | just 'non-standard' |
| 22:46.17 | De_Mon | bad bad bad |
| 22:46.22 | blitzrage | the joy of standards, is there are so many to choose from |
| 22:46.47 | marc7 | I'm invoking a call queue with exten => 2,2,Queue(Test|t|||45) -- if it isn't super visible from what i've typed in, the only options there are 't' to allow the call to be transferred, and have someone sit in the queue for 45 seconds... /// the queue has an rrmemory strategy, what I'm trying to figure out is how to have it go in a round-robin format... so each member has a penalty (member => SIP/john,1 / member => SIP/steve,2)... if john doesn't pi |
| 22:46.50 | *** part/#asterisk zerohalo (n=zeroHalo@pool-71-255-162-167.bstnma.east.verizon.net) |
| 22:46.51 | BiG^DoG | ok... just had a phone call and it was really scratchy sounding... is that more likely CPU or network? |
| 22:47.03 | *** join/#asterisk seanmh (i=fiber0pt@216.31.101.41) |
| 22:47.04 | De_Mon | * should raise a warning saying its non-standard and number should get depriciated by 1.8 |
| 22:47.06 | marc7 | oh right, and the timeout in the queue is 15 |
| 22:47.18 | De_Mon | my 2 cents |
| 22:47.37 | Grizzy | decremented : o ) |
| 22:48.39 | blitzrage | De_Mon: patches accepted |
| 22:49.38 | *** part/#asterisk cfh (n=luca@195.206.30.210) |
| 22:50.15 | marc7 | any ideas guys? I don't want to have to defer to using Agents instead of Members in the queue |
| 22:52.38 | BiG^DoG | says here jitter can cause scratchy voice calls |
| 22:52.43 | BiG^DoG | how do I tell if I have a jitter problem? |
| 22:54.07 | Greek-Boy | ManxPower: u saying record() is better than mixmonitor() ? |
| 22:55.07 | JT | they do totally different things |
| 22:55.17 | JT | record is not interchangeable with mixmonitor |
| 22:56.34 | Greek-Boy | does record give u two channels already mixed? |
| 22:56.44 | blitzrage | opposite of that |
| 22:56.45 | De_Mon | heh |
| 22:56.51 | blitzrage | it actually just records a single channel |
| 22:56.56 | Greek-Boy | ok |
| 22:57.02 | Greek-Boy | i thought monitor does that |
| 22:57.12 | Greek-Boy | anyway my mixmonitor is not working |
| 22:57.14 | blitzrage | MixMonitor() will take the two separate channels and mixes them together after call completion |
| 22:57.22 | blitzrage | Monitor() doesn't mix them and leaves them as two separate files |
| 22:57.26 | Greek-Boy | in the CLI it works fine with the variables but no .wav's are actually produced |
| 22:57.43 | *** join/#asterisk mog (n=mog@c-71-207-231-41.hsd1.al.comcast.net) |
| 22:57.43 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:57.54 | blitzrage | omgmoghikthxbye! |
| 22:58.43 | Qwell | zmog! |
| 22:58.46 | Qwell | :D |
| 22:58.48 | blitzrage | heh |
| 22:59.00 | blitzrage | wow... 6pm and still nothing on TV |
| 22:59.50 | *** join/#asterisk dlynes_home (n=dlynes@d154-20-45-103.bchsia.telus.net) |
| 23:00.11 | Greek-Boy | according to TFOT 2nd edition if i dont specify a path but just a .wav with mixmonitor it will use the "monitor" directory set in asterisk.conf. do they mean the spool directory set in asterisk.conf? |
| 23:00.23 | *** part/#asterisk waverly360 (n=waverly@ns2.dalcon.com) |
| 23:00.51 | blitzrage | /var/spool/asterisk/monitor I believe |
| 23:00.58 | blitzrage | or recordings... I forget |
| 23:01.22 | marc7 | hey guys, sorry to impose, but I'm certain this is an easy answer that I just can't find in any online documentation. How do you configure an "rrmemory" queue to ring the second member if the first member doesn't pick up? |
| 23:01.23 | blitzrage | ya -- monitor |
| 23:01.44 | blitzrage | marc7: it automatically falls through when the first member timeout expires |
| 23:02.09 | blitzrage | the timeout in Queue() is an absolute timeout I believe -- not per agent |
| 23:02.44 | marc7 | blitzrage: perfect! I had figured the Queue() timeout is absolute... so the first member timeout is defined by the "timeout" option in the [Test] section of queues.conf, right? |
| 23:02.55 | blitzrage | something like that |
| 23:03.51 | marc7 | OH! maybe in rrmemory I shouldn't be putting penalties on the other users |
| 23:04.15 | blitzrage | keep it simple at first -- then add features |
| 23:04.21 | Greek-Boy | blitzrage: so I have to set a monitor dir in asterisk.conf or will the spool dir setting take care of it? |
| 23:04.29 | marc7 | right, well I'm trying to make sure that the first person in the queue *always* gets called first |
| 23:04.42 | blitzrage | if you don't specify a directory, it'll use /var/spool/asterisk/monitor (by default) |
| 23:04.43 | marc7 | and I'm not entirely sure that's the behavior of rrmemory |
| 23:04.53 | blitzrage | marc7: that's not rrmemory |
| 23:04.57 | *** join/#asterisk techie (n=techie@adsl-76-214-9-1.dsl.lsan03.sbcglobal.net) |
| 23:05.02 | marc7 | blitzrage: what *is* that then? |
| 23:05.04 | blitzrage | rrmemory remembers the last position |
| 23:05.15 | blitzrage | so if person 3 answers, then the next call starts at person 4 |
| 23:05.22 | marc7 | has roundrobin been completely deprecated? |
| 23:05.23 | blitzrage | you want "roundrobin" to start at the top of the same list each time |
| 23:05.27 | blitzrage | and yes -- it has been |
| 23:05.51 | marc7 | so what the hell :D |
| 23:06.15 | marc7 | what can I use instead? |
| 23:06.41 | blitzrage | using 1.4? roundrobin still works |
| 23:07.55 | marc7 | but hasn't it been deprecated? what's taking its place? |
| 23:08.09 | Qwell | marc7: rrmemory |
| 23:08.45 | blitzrage | Qwell: I still don't understand why roundrobin was removed -- rrmemory and roundrobin act differently |
| 23:08.55 | blitzrage | marc7: deprecated doesn't mean it doesn't work |
| 23:08.55 | marc7 | Qwell: but how do I get the behavior of roundrobin in rrmemory? this is one thing that is completely beyond me... why deprecate that feature when it's obviously something people use |
| 23:08.57 | Qwell | dunno, Kevin did it |
| 23:09.00 | Qwell | I think |
| 23:09.24 | blitzrage | Qwell: ya, I think so -- I'm still not sure I agree with the decision, but if I really don't like it, I'll bring it up on the mailing list :) |
| 23:09.26 | marc7 | I can use penalties to have rrmemory target specific people first... but it keeps calling the same person even if they don't pick up |
| 23:09.36 | marc7 | blitzrage: asterisk-users? |
| 23:09.46 | *** join/#asterisk Bhaal (i=bhaal@freenode/unconfirmed/bhaal) |
| 23:10.12 | blitzrage | that'd probably be most appropriate after you verify it actually was removed in trunk -- there might be documentation of how to make rrmemory work the way you expect -- I've not looked |
| 23:10.25 | blitzrage | I don't subscribe to asterisk-users, which is probably why I haven't posted anything there |
| 23:11.20 | Qwell | The holiday month of November shall henceforth be called "Thanksgivemepresents". |
| 23:15.09 | *** join/#asterisk mamep (i=fallen@helios.edu.uoc.gr) |
| 23:15.21 | putnopvut | marc7: the penalty for a queue member is only invoked if the queue member is determined to be unreachable. Not answering doesn't satisfy that criteria. |
| 23:15.23 | *** join/#asterisk Bhaal (i=bhaal@freenode/unconfirmed/bhaal) |
| 23:16.06 | Qwell | ugh, not again |
| 23:16.13 | mamep | can someone help me with ooh323? |
| 23:16.27 | marc7 | putnopvut: I just figured that out through trial and error. anything you could suggest for me to accomplish the same task? voip-info.org has stated flatly that circular call distribution has gone the way of the dinosaurs |
| 23:16.52 | putnopvut | marc7, yeah unfortunately without some dialplan magic, you won't be able to do it in 1.4 |
| 23:17.15 | putnopvut | In trunk, however, there is a "linear" strategy which should do what you want (call members in the order they're listed in queues.conf) |
| 23:17.23 | mamep | anyone? |
| 23:17.34 | marc7 | awesome. I'm glad to hear that's coming. |
| 23:18.56 | blitzrage | putnopvut: aha -- good to know |
| 23:19.08 | JT | mamep: didn't you get the message last time? :) |
| 23:19.19 | mamep | jt : nah |
| 23:19.36 | mamep | they said me to get another solution |
| 23:19.42 | JT | mamep: it is very unlikely there will ever be anyone in here to help you |
| 23:19.47 | mamep | but i've managed to work with ooh323 |
| 23:19.51 | mamep | to get incoming calls |
| 23:19.58 | mamep | but having some problem with outgoin |
| 23:20.01 | JT | mamep: and chan_woomera seems to be the best H.323 option |
| 23:20.07 | JT | and we can help you how? :) |
| 23:20.27 | mamep | jt : first of all chan_woomera is not building for me.. |
| 23:20.38 | mamep | it tries to connect to a cvs which is down.. |
| 23:21.03 | mamep | that's why i tried to solve it with ooh323 |
| 23:21.46 | JT | and we can help you with it how? |
| 23:22.09 | mamep | maybe someone which has buiild chan_woomera |
| 23:22.35 | mamep | and i think is a problem with the audio codecs |
| 23:23.09 | mamep | cause i've managed to call using ooh323 but when i answer the call tha caller cant see it.. |
| 23:23.14 | mamep | and i get a wired message |
| 23:25.04 | mamep | maybe you can help me with this |
| 23:26.04 | *** join/#asterisk lowlevel (n=Stuart@CPE0017f2e2a3e8-CM000f9f7d6742.cpe.net.cable.rogers.com) |
| 23:26.09 | *** join/#asterisk coppice (n=chatzill@102.204.17.210.dyn.pacific.net.hk) |
| 23:28.05 | *** join/#asterisk kanatu (n=tracker@71.237.140.3) |
| 23:29.56 | jameswf | jbot, common sense |
| 23:29.57 | jbot | common sense is, like, applying simple logic and everyday reasoning to a problem. Doing so prevents you from asking stupid questions. |
| 23:30.01 | hmmhesays | seriously the croc pot is an awesome cooking tool |
| 23:30.17 | jameswf | jbot, moron |
| 23:30.18 | jbot | moron is, like, someone that types long lines starting with "jbot" |
| 23:30.20 | coppice | and croc is delicious |
| 23:30.31 | jameswf | jbot, botsnack |
| 23:30.31 | jbot | thanks, jameswf |
| 23:30.34 | hmmhesays | I just sampled some soup I made last night |
| 23:30.42 | hmmhesays | delicious |
| 23:31.04 | coppice | croc soup is pretty good. its a very traditional chinese medicine |
| 23:31.23 | Qwell | reptile? no thanks |
| 23:31.24 | hmmhesays | I may try chicken next time though, I used roast this time |
| 23:31.32 | Qwell | (crocs are reptiles, right?) |
| 23:31.47 | coppice | frog tastes very much like chinese. try that for a change |
| 23:31.58 | coppice | s/chinese/chicken - oops |
| 23:32.00 | hmmhesays | i don't know where to get that around here |
| 23:32.05 | Qwell | coppice: I was gonna say... |
| 23:32.11 | hmmhesays | I can get camel for sure |
| 23:32.15 | coppice | we get it from the supermarket |
| 23:32.29 | hmmhesays | I live in fargo |
| 23:32.30 | Qwell | coppice: chinese, or frog? |
| 23:33.08 | coppice | I keep several chinese at home, so I only need to go to the supermarket for frogs and crocodiles |
| 23:33.12 | mamep | JT : http://pastebin.ca/784983 |
| 23:33.14 | mamep | check this |
| 23:33.21 | mamep | i have a strange problem with cdr |
| 23:33.45 | mamep | also when i call a number it ring and if the calle answer the caller continues ringing.. |
| 23:34.50 | coppice | Qwell: you seem to have a reptile prejudice. if someone served you fron without saying, you probably wouldn't realise it was not chicken |
| 23:35.14 | Qwell | coppice: taste isn't the only thing I'm concerned about |
| 23:35.37 | Qwell | in fact - I was once offered chicken feet. CLEARLY, it would have tasted like chicken. |
| 23:35.49 | lowlevel | you must eat chicken feet |
| 23:35.54 | coppice | chickens feet are good |
| 23:35.55 | JT | Qwell: crocodile is really nice |
| 23:35.59 | Qwell | no, really, I mustn't |
| 23:36.01 | lowlevel | ;) |
| 23:36.11 | JT | mamep: maybe you don't get it.. we can't help you with it |
| 23:36.15 | lowlevel | this guy I used to work with used to say that to me all the time ... ' you must eat chicken feet' |
| 23:36.23 | lowlevel | apparently his father made him eat it often. |
| 23:36.30 | Qwell | lowlevel: the people I was with loved it |
| 23:36.45 | Qwell | they also got a few other "bizarre" foods |
| 23:36.46 | lowlevel | heh ;) |
| 23:36.56 | Qwell | blood pudding, for example |
| 23:37.04 | hmmhesays | you can watch mike rowe eat all kinds of nasty sh1t on dirty jobs |
| 23:37.09 | coppice | chickens feet sounds odd, but if you try them you'll probably find them rather yummy. I like the chillied |
| 23:37.16 | Qwell | I actually tried that - it wasn't good at all |
| 23:37.27 | Qwell | coppice: I'm sure they are good - I just don't really see the point |
| 23:37.48 | coppice | because they taste good. what other point would there be? |
| 23:37.59 | Qwell | there are things that taste far better ;) |
| 23:38.12 | Qwell | and don't require me sucking on a chickens foot |
| 23:38.27 | lowlevel | atleast it comes with a toothpick |
| 23:38.34 | lowlevel | or.. some toothpicks |
| 23:39.00 | coppice | blood pudding is more of an acquired taste. blood sausage is pretty common in europe, and blood pudding does exist. for some reason most people ignore that, and think its very asian |
| 23:39.21 | Qwell | coppice: it seemed pretty tasteless to me |
| 23:39.36 | Qwell | the blood pudding, that is |
| 23:39.46 | coppice | yeah, but the texture is not well liked by a lot who try it |
| 23:39.51 | Qwell | and I'd be cool with blood sausage - at least that is...food |
| 23:39.59 | Qwell | blood pudding, on the other hand, is just...blood |
| 23:40.35 | coppice | chickens feet are food. not especially healthy food, as they are mostly fat, but food nonetheless |
| 23:40.36 | *** join/#asterisk Defraz (n=t0tal@24-116-152-177.cpe.cableone.net) |
| 23:40.44 | Qwell | sure |
| 23:40.53 | Qwell | depends on your definition of "food" though :) |
| 23:41.08 | Qwell | cow tongue is "food" too, but I'm not gonna eat that either :D |
| 23:41.29 | moy | \quit |
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| 23:43.02 | Qwell | coppice: It seems like just a cultural thing. I'm sure there are things we eat that would be considered "bad" to others |
| 23:43.28 | coppice | I think most of what americans eat is considered bad by others |
| 23:43.35 | Qwell | heh |
| 23:44.06 | Qwell | I'm sure chicken feet are a heck of a lot better for you than most of the stuff you get from mcdonalds |
| 23:44.18 | coppice | you guys make sure the fat content of most meals exceeds chicken's feet. we don't make every meal from that stuff |
| 23:44.26 | Qwell | yeah |
| 23:44.34 | moy | Qwell: cow tongue is delicious |
| 23:44.43 | Qwell | moy: I'm not disputing that :) |
| 23:44.44 | rob0 | If those feet were so good, why didn't the poor chicken use them to get away? |
| 23:44.57 | moy | :P you should give it a try |
| 23:45.42 | coppice | rob0: it sounds like the old genetically engineered 3 legged chicken joke is coming up |
| 23:46.00 | Qwell | 3 legged chicken? So, you've been to KFC? |
| 23:46.27 | *** join/#asterisk Navion (n=billp@75-105-41-123.cust.wildblue.net) |
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| 23:46.37 | coppice | Qwell: you're not a vegetarian, are you? |
| 23:46.43 | Qwell | not at all |
| 23:47.04 | rob0 | I'm a humanitarian, myself. |
| 23:47.06 | kanatu | Can anyone help me with getting blind transfer to work on asterisk 1.2.21.1? |
| 23:47.33 | coppice | not at all might be an overstatement considering this conversation |
| 23:47.35 | Qwell | coppice: though, I've been to a vegetarian chinese place...it was pretty awesome |
| 23:47.53 | coppice | you mean the fake meat places? |
| 23:47.56 | Qwell | yeah |
| 23:48.08 | Qwell | is that common over there at all? |
| 23:48.38 | coppice | I think that is a cop out. if you don't want to eat meat, why not eat veggie that actually looks like veggie. |
| 23:48.46 | Qwell | coppice: no idea |
| 23:48.59 | Qwell | but, every time I went, there were Buddhists and such there |
| 23:49.01 | Navion | Help setting up Sangoma A200 FXO cards? |
| 23:49.15 | Qwell | erm, like |
| 23:49.18 | Qwell | you know |
| 23:49.37 | Qwell | monks, I guess? |
| 23:50.17 | coppice | there are a number of buddist restaurants that serve that kind of food, and the monsatery here http://en.wikipedia.org/wiki/Tian_Tan_Buddha serves that to many tourists each day |
| 23:50.56 | Qwell | interesting |
| 23:57.28 | *** join/#asterisk BBHoss (n=hoss@146.229.191.76) |
| 23:57.43 | coppice | Qwell: you sound like the sort of person who'd be real fun at a chinese banquet :-) |
| 23:59.21 | Qwell | coppice: yeah... |
| 23:59.39 | Qwell | I went to a dimsum(sp) place with coworkers one time. They weren't very happy with me :p |