| 00:00.04 | *** join/#asterisk Dr-Linux|home (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 00:00.04 | ManxPower | since there were no actual dialed digits |
| 00:00.05 | litage | ManxPower: ah, yes |
| 00:00.05 | hrmphh | i.e. USERS=${USERA}&${USERB} instead of USERS=SIP/usera&SIP/userb |
| 00:00.09 | ManxPower | litage: do you want the actual dialed number in the CDR? |
| 00:00.25 | Dr-Linux|home | i'm working on a perl script which will montor my ATAs and PRI's, but having some problem with CGI |
| 00:00.34 | Dr-Linux|home | anybody knows CGI/Perl? |
| 00:00.39 | ManxPower | if so in the macro do exten => s,1,Goto(${MACRO_EXTEN},1) then have an exten => X.,1,rest of your macro |
| 00:00.42 | litage | ManxPower: yes, unless there are strong enough reasons not to do so |
| 00:00.58 | ManxPower | ...er.. |
| 00:01.10 | ManxPower | exten => _X.,1,rest of your macro |
| 00:01.19 | litage | Dr-Linux|home: that's too general of a question. ask a real question |
| 00:01.45 | *** part/#asterisk lachesis (n=lachesis@mon75-2-81-57-110-215.fbx.proxad.net) |
| 00:01.46 | ManxPower | litage: want a file that contains several macros, some very complex, some very simple? |
| 00:02.07 | ManxPower | http://www.fnords.org/~eric/macros.inc |
| 00:02.11 | Dr-Linux|home | litage: well, on the command line, output is just fine, but on the web it display nothing |
| 00:02.18 | litage | ManxPower: thanks, i'll have a look through it |
| 00:02.21 | Dr-Linux|home | litage: lemme give you URL for my script |
| 00:03.07 | litage | Dr-Linux|home: i don't mean to be rude, but my recommendation on how to ask a question didn't imply that i can help with your question |
| 00:03.52 | Dr-Linux|home | litage: i'm sorry, i don't know proper manners in english |
| 00:06.03 | Dr-Linux|home | litage: also i was little confused asking cgi/perl question here |
| 00:06.37 | hrmphh | nobody knows why those variables arent interpolted? |
| 00:06.40 | hrmphh | interpolated rather? |
| 00:06.41 | hrmphh | urgh |
| 00:07.09 | litage | Dr-Linux|home: not a problem. in the future, ask direct questions rather than "does anyone know X?" or "is anybody using Y?". also, if your problem is a purely perl problem, #perl is the right channel to ask |
| 00:07.40 | Dr-Linux|home | cool |
| 00:07.46 | litage | Dr-Linux|home: gl |
| 00:08.36 | Dr-Linux|home | litage: actually it's odd problem, bcoz perl guys may recomment asterisk channel, since asterisk stuff are there :P |
| 00:08.39 | Dr-Linux|home | anyway .. thanks |
| 00:08.57 | mmartinn | Dr-Linux|home: Do you have anything working yet? |
| 00:09.26 | Dr-Linux|home | mmartinn: hey there :) |
| 00:09.45 | Dr-Linux|home | mmartinn: i'm so closed, looks little problem, but only on the cgi/web |
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| 00:22.33 | *** join/#asterisk diclophis-work (n=jbardin@65.203.37.58) |
| 00:22.35 | diclophis-work | hello all |
| 00:22.38 | flenders | joobie: still there? |
| 00:22.46 | flenders | sorry mate, was on a meeting |
| 00:23.07 | diclophis-work | can anyone give me a round number on the cost of a enterprise grade sip proxy device, that will take a t1 and convert it to a voip trunk to be delivered to asterisk |
| 00:24.19 | gambolputty | $1 |
| 00:24.21 | mmartinn | A sip proxy wouldn't deliver a voip trunk to Asterisk from a T1... it would deliver SIP channels, no? |
| 00:24.40 | mmartinn | If it was truly converting T1 -> SIP |
| 00:25.17 | ManxPower | mmartinn: A SIP proxy is program that lets you manage SIP packets. It does not let you manage audio packets (that uses RTP protocol for SIP), it does nothing else |
| 00:25.32 | mmartinn | ManxPower: exactly, so he isn't looking for a sip proxy |
| 00:25.50 | mmartinn | Perhaps you mean something like http://www.red-fone.com/? |
| 00:25.55 | mmartinn | (to diclophis-work ) |
| 00:26.25 | ManxPower | diclophis-work: you are looking for a SIP "gateway" or "media converter" |
| 00:27.02 | mmartinn | ManxPower: oh, I guess they do make those... fancy... |
| 00:27.16 | mmartinn | ManxPower: When I thought of SIP Gateway, I thought "you need Asterisk" |
| 00:27.30 | mmartinn | I didn't know there was hardware out there to do it... hmmm |
| 00:27.33 | ManxPower | mmartinn: Asterisk makes a pretty good PSTN/SIP gateway |
| 00:28.59 | ManxPower | Many people use old Ascend boxes, some of them support SIP/PSTN gateway funtions |
| 00:29.29 | Innatech | I'm a little thick this afternoon. Why not bring the T1 right into the * box? |
| 00:29.59 | ManxPower | Innatech: Maybe you need a device that supports higher speed ports than T-1. |
| 00:30.17 | ManxPower | Or if you need to do large amount of PSTN/G729 conversion |
| 00:30.22 | Innatech | aha. >bonk< Thanks. |
| 00:30.54 | drfreeze | Any Grandstream experts out there know if the butdgetone 100 is remotely configurable like the polycom phones? |
| 00:31.29 | ManxPower | drfreeze: The Grandstream BT100/BT101 is what they put in Dante's hell for use by telephone guys that were very very very bad |
| 00:32.38 | drfreeze | ManxPower: I have a client deploying 80-90 phones - I told him today he should get 501s and 301s. He said he had just bought 50 BT's for $30 a piece. |
| 00:32.51 | drfreeze | I told him to quickly sell them. :) |
| 00:33.20 | ManxPower | drfreeze: soon the client will realize WHY they were $30 |
| 00:33.28 | drfreeze | ManxPower: :) lol |
| 00:33.39 | drfreeze | ManxPower: so, are you saying the answer is NO? |
| 00:34.14 | ManxPower | drfreeze: I would suggest looking at the grandstream support forums and/or voip-info.org. |
| 00:34.24 | drfreeze | k |
| 00:34.33 | ManxPower | You realize that the BT10x phones only have a number display, not an alpha display, right? |
| 00:34.49 | drfreeze | doesn't really matter |
| 00:34.59 | drfreeze | I would prefer a phone that is backlit tho |
| 00:35.05 | ManxPower | and there is no BT100, there is the 101 and the 102 |
| 00:35.14 | drfreeze | Does polycom have any backlit phones? |
| 00:35.25 | ManxPower | both products only support 10Mbps |
| 00:35.26 | drfreeze | then it is a 101. |
| 00:35.34 | drfreeze | what! |
| 00:35.58 | drfreeze | would that limit an 80 phone roll out? |
| 00:36.09 | ManxPower | drfreeze: they also tend to lock up at random times and in the past 5 years I don't think GS has ever released stable firmware for those phones. |
| 00:36.24 | ManxPower | drfreeze: only if you are stupid enough to put them on a hub rather than a switch |
| 00:37.26 | drfreeze | Does polycom have any backlit phones? |
| 00:37.48 | ManxPower | drfreeze: I'll bet the secret web site www.polycom.com will have info about the Polycom phones. |
| 00:38.11 | drfreeze | ManxPower: where do you get those mystical powers? |
| 00:38.15 | ManxPower | drfreeze:the 30x, 50x, and 60x phones are not backlit. Their new models might. |
| 00:38.58 | ManxPower | drfreeze: Attending orgies with psychedelic drugs, of course. |
| 00:39.22 | Innatech | Manx: I have that cert. |
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| 00:45.15 | drfreeze | ManxPower: buggers. only 550 is backlit |
| 00:45.18 | drfreeze | not the 330 |
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| 00:52.30 | drfreeze | hmmm, front lit. Not sure how good that is |
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| 01:31.21 | jeffgus | question |
| 01:31.32 | Qwell | answer |
| 01:31.36 | jeffgus | can asterisk work as a h.323 to isdn gateway? |
| 01:31.42 | jeffgus | or h.323 to t1 gateway? |
| 01:31.52 | jeffgus | err... h.323 over IP |
| 01:32.06 | jeffgus | i see that it can do audio |
| 01:32.07 | jeffgus | but |
| 01:32.26 | jeffgus | what if i connect a Polycomm video system to it? |
| 01:33.18 | jeffgus | it seems to me that asterisk will pass call setup back and forth |
| 01:33.36 | jeffgus | but does it actually do the right thing with video? |
| 01:33.58 | hrmphh | why am i getting "unable to create channel of type 'Zap' (cause 0 - Unknown) |
| 01:34.03 | hrmphh | if i set TRUNK to be like Zap/4 it works fine |
| 01:34.06 | hrmphh | but when i try to make it a group |
| 01:34.15 | hrmphh | by doing Zap/g1 |
| 01:34.17 | hrmphh | it fails :( |
| 01:35.31 | BrianHV | are there providers in the US besides vonage that will give me a phone number that will ring an asterisk setup? |
| 01:35.34 | hrmphh | nm had group in the wrong place |
| 01:36.56 | generalhan | hmm |
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| 01:41.50 | hrmphh | can someone recommend a softphone w/transfer? |
| 01:41.58 | hrmphh | apprently x-lite makes you pay for eyebeam 1.5 to transfer? |
| 01:42.00 | hrmphh | or is there a workaround? |
| 01:43.39 | Qwell | a workaround to paying? |
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| 01:45.40 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 01:47.36 | hrmphh | no like |
| 01:47.45 | hrmphh | another way instead of using the transfer button |
| 01:47.46 | hrmphh | flash or something |
| 01:48.20 | *** part/#asterisk kieran491 (n=ehh@c211-28-159-50.brasd1.vic.optusnet.com.au) |
| 01:51.49 | hrmphh | can someone recommend a free softphone that allows transfer? |
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| 01:54.49 | litage | hrmphh: we heard you the first time. you don't need to repeat yourself |
| 01:54.56 | litage | hrmphh: try sjphone |
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| 01:55.44 | *** join/#asterisk p0g0 (n=pogo@madwifi/support/p0g0) |
| 01:55.50 | hrmphh | trying |
| 01:57.16 | piper69 | is there is free PSTN |
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| 01:58.05 | flenders | piper69: free PSTN? you mean a free analogue line? |
| 01:58.53 | flenders | or free calls to the PSTN? |
| 01:59.51 | piper69 | flenders: so far what i understand "correct me if i am wrong please", if i got my Asterisk box running i will only be able to call people PBX |
| 02:00.41 | flenders | piper69: if you get an asterisk box running, you can ring anyone you want |
| 02:00.52 | piper69 | beside my asterisk what do i need to call someone let say in australia for example? |
| 02:01.15 | flenders | piper69: a landline or a voip provider |
| 02:02.23 | *** join/#asterisk JT (n=jon@unaffiliated/jt) |
| 02:02.49 | piper69 | flenders: i just finished chapter 2 of asterisktfot, and trust me if i wanted to use the landline if would've not bother to read about asterisk, i am trying to learn |
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| 02:04.24 | piper69 | flenders: ber_ was telling me there is free voip providers |
| 02:04.42 | flenders | piper69: FWD? |
| 02:05.00 | flenders | piper69: not sure if you can call landlines for free though |
| 02:05.09 | piper69 | ~FWD |
| 02:05.21 | jbot | extra, extra, read all about it, fwd is Free World Dialup: Brainchild of Jeff Pulver. URL: http://www.pulver.com/fwd/ |
| 02:07.01 | piper69 | flenders: yes FWD, what does it mean then free world dialup |
| 02:07.01 | JT | it's just a service to talk to other people using fwd |
| 02:07.02 | flenders | piper69: I thought you could only talk to people on FWD |
| 02:07.02 | flenders | piper69: like a skype, but using SIP or IAX |
| 02:07.24 | flenders | piper69: it would be the same as adding skype to asterisk, i'd guess |
| 02:07.29 | flenders | if you could |
| 02:09.23 | mihinomenest | is it just me or are the freepbx screenshots broken? |
| 02:09.23 | piper69 | flenders: can you clear it up please, can i or i can't? |
| 02:09.49 | JT | lol freepbx |
| 02:10.19 | mihinomenest | bossman wants a webterface. |
| 02:10.32 | mihinomenest | I don't see one in the base asterisk package. |
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| 02:16.22 | flenders | piper69: you might want to read the stuff on their website |
| 02:16.37 | flenders | piper69: I'm pretty sure it's better explained there |
| 02:21.03 | flenders | oI? over Internet? |
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| 02:48.10 | Chr1z28 | If I want to use a central asterisk box for multiple branch offices all of whom will use sip phones, is a digium card or the like still required? |
| 02:48.48 | *** join/#asterisk ivanfm_ (n=ivanfm@c93481ec.virtua.com.br) |
| 02:51.07 | hrmphh | can someone please take a look at http://pastebin.ca/423359 and shed some light on why i dont hear any audio when dialing/apparently connected? using an analog phone plugged in to the FXS port ... works fine receiving calls |
| 02:51.30 | hrmphh | also the remote end never receives the call from the handset ... ive tried dialing my cell |
| 02:53.23 | hrmphh | nm L1 problem :) |
| 03:02.23 | Chr1z28 | can I use sip phones w/o having to have a digium card? |
| 03:02.58 | JunK-Y | Chr1z28: yes ya can. |
| 03:03.42 | Chr1z28 | JunK-Y: ok..so if I have remote locations they can use sip phones from there and hit my asterisk setup w/o me needing any special hardware (ie. digium card etc) ? |
| 03:04.11 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 03:07.47 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 03:12.19 | ber_ | is there any way to play wav/mp3 from an offset in asterisk |
| 03:13.44 | *** join/#asterisk maverickbna (i=sentinel@wikipedia/Shadowhntr) |
| 03:15.07 | hrmphh | is there a default m usic on hold that just plays tones or something? |
| 03:15.10 | hrmphh | rather than actual music? |
| 03:15.14 | hrmphh | like occasional beeps? |
| 03:19.03 | *** join/#asterisk yxa (n=lonari@58.185.90.101) |
| 03:20.54 | FuriousGeorge | hrmphh: make a one minute mp3 that starts with a beep |
| 03:21.06 | *** join/#asterisk sysreq (n=sysreq@modemcable171.134-81-70.mc.videotron.ca) |
| 03:21.09 | FuriousGeorge | or |
| 03:21.42 | FuriousGeorge | have the beep.wav file provided with asterisk in the moh dir, along with a one minute mp3 of dead air |
| 03:21.44 | JunK-Y | see the sounds beep ? |
| 03:22.09 | JunK-Y | personnaly i dont see or would like getting a 3 minutes long BEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEEPPPPPPPPPp |
| 03:22.13 | JunK-Y | heheh |
| 03:22.26 | FuriousGeorge | JunK-Y: he means an beep (minute of silence) beep |
| 03:22.31 | hrmphh | lol |
| 03:22.34 | hrmphh | just a sec of beep |
| 03:22.36 | hrmphh | then like 30seconds dead air |
| 03:22.40 | hrmphh | ill take that advice thanks |
| 03:22.48 | FuriousGeorge | np |
| 03:23.15 | JunK-Y | ha, just see the silence dir/ and beep sounds, with that ya will be able to tweak something |
| 03:24.00 | hrmphh | thanks |
| 03:24.07 | ber_ | i can slice the wav file myself to remove the first X minutes |
| 03:24.27 | ber_ | nm |
| 03:27.15 | *** part/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
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| 03:29.57 | *** join/#asterisk [tasty]freeze (n=simba@199.111.147.48) |
| 03:31.10 | [tasty]freeze | how are sip calls routed using asterisk? from one sip phone to another; im a little foggy on this are they passed thru the asterisk server or are they directly connected to each other |
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| 03:47.29 | sumasuma | How can I make the asterisk to behave with the codec preference of UA than the priority mentioned in the asterisk config ? |
| 03:48.18 | sumasuma | I mean if, A -> Asterisk -> B , i want the preference of the codec of A to be passed to B, but asterisk sends the preference as per the configuration of the channel file (ex., sip.conf) |
| 03:49.19 | sumasuma | Is there is anyway to take as per the clients preference ? |
| 03:50.34 | [tasty]freeze | install a laser on the top of your desk and enable the laser option in the asterisk config. |
| 03:50.53 | [tasty]freeze | if that doesnt work im not sure. |
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| 03:51.11 | sumasuma | [tasty]freeze: you worked before in bar as a server ? |
| 03:51.38 | [tasty]freeze | i am afraid not. i think im afraid. why? |
| 03:52.07 | [tasty]freeze | should i put it on my todo list? |
| 03:52.26 | sumasuma | [tasty]freeze: i have seen the laser work much on the bar and the servant in the bar used to on and off that one |
| 03:52.49 | sumasuma | thought you are much familier with that |
| 03:52.55 | [tasty]freeze | ah i see. i am not. |
| 03:53.20 | [tasty]freeze | i really must shower however because i smell; i wish you the best of luck finding your answers sumasuma |
| 03:53.34 | sumasuma | okie |
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| 04:19.00 | jql | any advice on what headset to get (or not) for a polycom 501? |
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| 04:20.13 | ManxPower | I never thought I'd say this, but I'm starting to hate analog lines |
| 04:21.08 | ManxPower | I have a 500hz - 650Hz hum (it fluctuates) |
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| 04:24.52 | aptura | odd |
| 04:25.26 | aptura | do you have the signal some kind of scope ? |
| 04:26.01 | ManxPower | Yes, a Sage Instruments 930A (it is a test set for phone lines) |
| 04:26.06 | aptura | jql plantronics |
| 04:26.20 | aptura | nice |
| 04:26.20 | ManxPower | The hum is at about -49db |
| 04:26.22 | aptura | pretty low |
| 04:26.26 | ManxPower | Bought it for like $50 off of ebay |
| 04:26.34 | ManxPower | aptura: yes but users are still complaining |
| 04:26.55 | ManxPower | I need to test it at the demarc and see if I still have it. |
| 04:27.34 | ManxPower | We are 11 miles from the CO, but about 2 - 3 miles from this site is a SLIC96 between us and the CO |
| 04:27.36 | aptura | I have taken a couple years of electronics and cannot think of what discreet electronic componets would produce 500 hz unless it is a multiple of the fundemental frequency. Harmonics! :) |
| 04:27.59 | ManxPower | I suspect it is a harmonic of 60Hz |
| 04:28.01 | aptura | It could be some harmonics not being filtered out. |
| 04:28.53 | aptura | If there are electrolitic capacitors in the circuit thay could be getting dry and not working right but would not know what else it could be. Perhaps poor shielding from a harmonic source. |
| 04:29.26 | aptura | Actually it could not be electrolytic it would be disk capacitor. |
| 04:29.27 | ManxPower | Bell corrected a "ground fault" on one of the lines. |
| 04:30.02 | [TK]D-Fender | jql, Plantronics M12 amplifier & H261 binaural polaris headset |
| 04:30.03 | aptura | tiny disk capacitors are used between ground and branches in circuits mostly. |
| 04:30.41 | aptura | so what was the result? |
| 04:32.54 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 04:32.56 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 04:34.48 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 04:36.25 | *** join/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:39.49 | aptura | I once worked at Fluke some of there network/telco test gear was in the sub 15-20k rang for the upper end hand sets. |
| 04:40.22 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 04:40.31 | ManxPower | well that was fun. |
| 04:40.50 | ManxPower | the result was no hum when I listened to the line, but that was on a butt set, not a real phone |
| 04:41.02 | ManxPower | I get the same hum, but at about -62db at the demarc |
| 04:41.30 | aptura | butt set is one of those orange test phones? |
| 04:42.38 | aptura | disregard |
| 04:42.39 | aptura | :) |
| 04:43.14 | *** part/#asterisk [TK]D-Fender (n=Joe@64.235.216.2) |
| 04:44.12 | ManxPower | on the 2nd line I get a 120hz - 190hz at -59db |
| 04:44.33 | ltdwk | unlucky |
| 04:45.04 | aptura | anything at 180? |
| 04:45.45 | aptura | 120 I can understand but 190? |
| 04:47.09 | aptura | Manx looks like something I used 15 years ago but for Avionics. |
| 04:47.50 | aptura | The sage is not something you normally lug around |
| 04:53.02 | ManxPower | no it isn't |
| 04:53.05 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 04:53.26 | ManxPower | on hook it is around 80hz at -50db or so |
| 04:53.41 | aptura | dbm |
| 04:53.55 | Focher | anyone chance someone can recommend a service partner that knows Asterisk in Australia? |
| 04:54.25 | aptura | One of the really cool things I did in avionics was lug around a all in one test set that would transmit everything to ILS. |
| 04:54.54 | aptura | In the wrong hands it could cause a pilot to diviate from say his glide scope or localizer right into a building. |
| 04:55.33 | aptura | Focher not sure. |
| 04:55.40 | ltdwk | Focher: when you say service partner, what do you mean |
| 04:55.59 | Focher | consulting for setting up |
| 04:56.12 | ltdwk | Focher: Where are you located? |
| 04:56.19 | Focher | sydney |
| 04:58.51 | ManxPower | I wish this sage box did voltage measurements |
| 05:01.44 | flenders | focher: I know asterisk |
| 05:01.55 | flenders | and I'm in sydney too |
| 05:01.55 | flenders | :D |
| 05:01.59 | ltdwk | I know kung fu |
| 05:02.43 | flenders | I also know how to ride a bike |
| 05:03.17 | aptura | ManxPower probebly invest in a fluke DMM or small scope. |
| 05:04.45 | aptura | man its time to crash. I am writing my asterisk configs by hand and doing okay so far just trying to recall what application is usally included along side DigitTimout when a person is dialing a extention. |
| 05:05.07 | *** join/#asterisk hedge77 (n=ambray@user-0c8hj7t.cable.mindspring.com) |
| 05:05.19 | Focher | sorry but not seeing my msg windows... |
| 05:06.14 | ltdwk | i sent you a msg asking what the scope of your requirements are |
| 05:07.03 | Focher | I figured you sent something but I am trying to find where this IRC client put it.... |
| 05:07.34 | flenders | focher: which client is it? |
| 05:08.01 | Focher | iceirc |
| 05:08.04 | Focher | icechat |
| 05:09.23 | Focher | it's not in the console window and not in the chat window. Window menu doesn't show anything either |
| 05:10.27 | hedge77 | wow my cable is really screwy. modem is fine, channels above 100 are fine, but <100 is out |
| 05:10.31 | Focher | Sorry. I'll just write it here. We are really just evaluating Asterisk and potentially would have it in the Asia Pac region. |
| 05:11.54 | kavit | Focher: did you see my notice? |
| 05:12.05 | Focher | yeah. Will take a look |
| 05:12.06 | flenders | focher: kavit: he can't read pvt msg |
| 05:12.13 | flenders | kavit: he can't read pvt msg |
| 05:12.20 | kavit | flenders: i did a /notice :P |
| 05:12.21 | Focher | that worked, actually |
| 05:12.24 | flenders | ahh ok |
| 05:12.56 | kavit | Focher: We can help you scope the whole thing and we provide free pre sales consultation by phone or in person |
| 05:13.47 | kavit | Focher: we can even host a PBX for you so you can use it from pretty much anywhere |
| 05:13.48 | Focher | great. Will take a look at the website first. |
| 05:13.56 | kavit | Focher: no worries.... |
| 05:14.27 | Focher | How big have your deployments been? |
| 05:15.42 | kavit | Focher: the biggest one I have consulted on with Asterisk was over 3 cities with 50+ seats each... I have also done Cisco Call manager deployments with over 1000 users |
| 05:16.05 | kavit | Focher: I was a part of a team though... before I started this company... so I cant take all the credit for it.... |
| 05:16.12 | Focher | ever heard of Sphere? |
| 05:16.22 | kavit | Focher: Sphere IT? |
| 05:16.47 | Focher | naw, Sphere Communications. VOIP solution, but haven't seen them outside the US / Mexico |
| 05:17.16 | kavit | Focher: I have heard of them now :P |
| 05:17.22 | Focher | hehe |
| 05:17.23 | kavit | Focher: just googled them |
| 05:17.33 | Focher | We use them in the US. |
| 05:18.19 | *** join/#asterisk dlynes_laptop (n=dlynes@S01060016b6c052ee.vc.shawcable.net) |
| 05:18.33 | kavit | Focher: We are relatively new.. so not as big as them, but we provide a complete traditional telephony replacement with proper CLIP and anything else you may want |
| 05:19.39 | kavit | Focher: we do a lot of custom programming work around Asterisk and integration with business software as well and provide support. Good feedback from the market |
| 05:20.16 | Focher | So like integration with LDAP? |
| 05:20.40 | kavit | Focher: if the customer desires..... |
| 05:21.12 | *** join/#asterisk moranil (n=moranil@122.162.69.132) |
| 05:21.43 | ltdwk | LDAP in asterisk is not the most stable thing at the moment |
| 05:22.27 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-75-1-17-69.dsl.hstntx.sbcglobal.net) |
| 05:22.28 | Focher | I am not claiming expertise with Asterisk, but my brief checking of it is that the backend configuration tools and methods leave a bit to be desired in an enterprise environment |
| 05:22.57 | kavit | Focher: depends on how extensive your requirements are.... |
| 05:23.55 | kavit | Focher: it is a trade off... I guess... you get a lot with asterisk that is hard to come by elsewhere |
| 05:24.14 | Focher | well, we pretty much store everything regarding a person's identity in our eDirectory structure so we would require that level of integration |
| 05:24.39 | Focher | no reason |
| 05:25.01 | kavit | Focher: a trial can always be set up to evaluate if it meets your needs |
| 05:25.03 | ltdwk | I've integrated our * into LDAP here as best I can |
| 05:25.50 | Focher | we also sometimes just sync the identity information to an external directory. Identity Manager let's you do it. |
| 05:26.00 | kavit | ltdwk: what has been the biggest gripe/caveat? |
| 05:26.47 | ltdwk | it works fairly well, i'm pretty much 100% using it for voicemail |
| 05:26.50 | aptura | night |
| 05:27.00 | ltdwk | using the realtime ldap code |
| 05:27.58 | ltdwk | the biggest gripe was probably that the way I set my stuff up, the voicemail users don't have a context |
| 05:28.54 | ltdwk | asterisk had some pieces of code that didn't work properly if there was no context |
| 05:29.02 | ltdwk | only one really... |
| 05:29.07 | ltdwk | but i submitted a patch to fix that |
| 05:30.34 | ltdwk | I added in some dialplan stuff to look up caller ID names using LDAPget, too, but I found that LDAPget was unstable. -- it caused * to segfault. |
| 05:30.43 | *** join/#asterisk Chris-NB (n=chris@ip.tech.t-mobile.at) |
| 05:31.04 | aptura | did u debug it? |
| 05:31.22 | ltdwk | couldn't, it was a live system |
| 05:31.39 | aptura | how many users |
| 05:31.58 | ltdwk | around 150 voicemail users |
| 05:32.07 | aptura | k |
| 05:32.09 | *** join/#asterisk andrew` (n=andrew@69-12-140-101.dsl.dynamic.sonic.net) |
| 05:32.21 | aptura | well if you have a identical server use that to test and debug. |
| 05:32.21 | ltdwk | it would work fine for a few queries, then die |
| 05:33.00 | ltdwk | when I get some spare time, I'll do some testing late in the evening and see if I can debug it |
| 05:45.22 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 05:53.10 | kavit | ah I just got an email asking if I could do work with Nortel MPS/VPS |
| 05:53.16 | kavit | heh good old days |
| 05:53.33 | kavit | didn't know anyone still used VPS |
| 05:54.27 | ltdwk | never heard of it |
| 05:55.19 | kavit | ltdwk: its a beast of a system... comes with its own rack and chassis... used by telcos here in australia for ivr and speech recognition |
| 05:55.35 | kavit | was used I should say |
| 05:58.47 | *** join/#asterisk zeeesh (i=GrTB0T@202.38.55.125) |
| 05:58.48 | zeeesh | hi |
| 05:59.22 | *** part/#asterisk moranil (n=moranil@122.162.69.132) |
| 06:04.18 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 06:13.54 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:24.32 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 06:27.00 | *** join/#asterisk rward (n=rward@dsl-241-70-69.telkomadsl.co.za) |
| 06:27.31 | rward | I'm having problems getting DTMF and "*1" recording working correctly |
| 06:27.38 | rward | anyone have any experience with it? |
| 06:28.03 | rward | DTMF from my IAX and SIP phones is getting "picked up" as a *, then a 1 |
| 06:28.21 | rward | when I use a handset it gets picked up as a "*1" and works correctly |
| 06:29.08 | rward | Using AsteriskNOW and a TDM2400 card |
| 06:30.01 | *** join/#asterisk lfurrea (n=lfurrea@ip201-31-10-190.ct.co.cr) |
| 06:31.30 | lfurrea | anyone knows if Digium's new VoiceBus technology would make the cards compatible with PCI Express? |
| 06:37.28 | *** join/#asterisk newsmafia (n=newsmafi@wsip-68-224-174-204.sd.sd.cox.net) |
| 06:38.02 | *** part/#asterisk rward (n=rward@dsl-241-70-69.telkomadsl.co.za) |
| 06:45.31 | sumasuma | how to use refer (sip) in asterisk ? |
| 06:47.29 | *** join/#asterisk shodan- (n=shodan@ip216.99-113-216.pppoe4.joliette.intermonde.net) |
| 06:50.30 | *** join/#asterisk d00gster (n=doughant@bas1-toronto12-1088929080.dsl.bell.ca) |
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| 07:04.44 | *** join/#asterisk jjshoe (n=jjshoe@cpe-76-175-157-242.socal.res.rr.com) |
| 07:05.11 | jjshoe | does * support, and/or what's the name of a transfer where the transfering party exits leaving the two remotes connected? I'm not looking to use centrex xfer and clear |
| 07:12.35 | *** join/#asterisk felipex (n=dsfdsf@88-149-172-71.f5.ngi.it) |
| 07:19.03 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 07:21.53 | *** part/#asterisk ikaRus1 (i=none@80.179.36.48.static.012.net.il) |
| 07:23.15 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 07:28.19 | fetcher | jjshoe: with ISDN, if you want to hand the call back to your upstream switch (not consuming any B channels post-transfer), that's called variously 2BCT, TBCT, or ECT |
| 07:29.09 | fetcher | jjshoe: telco has to provision your service to allow for this also, often at extra cost |
| 07:29.18 | fetcher | jjshoe: not sure about the state of Asterisk support... |
| 07:34.00 | *** part/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 07:40.16 | *** join/#asterisk Jynger (n=rip@tti.tt.ee) |
| 07:40.32 | Jynger | hi |
| 07:41.09 | Jynger | does Asterisk 1.2.1 support t38 pass through ? |
| 07:41.17 | Jynger | 1.2.17 |
| 07:43.38 | fetcher | Jynger: Unfortunately not, as far as I know. 1.4.x might have it |
| 07:45.55 | Jynger | aha, also for it to work my iax2 trunk provider must also have 1.4.x with t38 pass through ? |
| 07:46.53 | Jynger | and the other between the 2 t38 ata's ? |
| 07:47.52 | tzafrir | Jynger, no, even T.38 passthrough is only supported with Asterisk 1.4.x |
| 07:47.56 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 07:47.59 | tzafrir | not in 1.2 |
| 07:47.59 | *** join/#asterisk felipex (n=dsfdsf@88-149-172-71.f5.ngi.it) |
| 07:49.03 | Jynger | hmm, i wonder why they still use 1.2 then |
| 07:51.12 | tzafrir | 1.4.2 caused a number of unexplained 100% CPU states on my laptop. I had to drop -p |
| 07:51.53 | tzafrir | of course this is probably an issue with my specific configuration and such. |
| 07:52.09 | *** join/#asterisk nfi|ermes (n=ermes@217.220.121.62) |
| 07:52.16 | tzafrir | But I'm still not as inclined to use it for production. |
| 07:52.28 | jjshoe | fetcher not using termination points, on asterisk itself |
| 07:52.30 | nfi|ermes | hi all |
| 07:52.35 | jjshoe | like putting everyone in a meetmee basically |
| 07:52.44 | Jynger | nja, but i mean, if i set up 1.4.2 pbx and want my users to be able fax over t38 it still wont work as our provider has 1.2.x that knows nothing about t38 ? |
| 07:52.51 | nfi|ermes | anyone can help me to understand how to add an application ??? |
| 07:53.28 | nfi|ermes | i d like to add http://www.pbxfreeware.org/archives/2005/06/new_download_--.html to my asterisk 1.4 |
| 07:53.47 | *** join/#asterisk IgorG (n=FeedomPa@195.162.32.126) |
| 07:54.10 | nfi|ermes | should i comple app_intercept.c ?? then ? |
| 07:56.50 | Jynger | is that even asterisk ? |
| 07:57.26 | *** join/#asterisk darkskiez (i=mhb@bb-87-81-62-203.ukonline.co.uk) |
| 07:58.08 | *** join/#asterisk friedrich| (n=friedric@e177255054.adsl.alicedsl.de) |
| 08:01.26 | *** join/#asterisk CrazyTux[m] (n=CrazyTux@adsl-75-1-17-69.dsl.hstntx.sbcglobal.net) |
| 08:02.17 | *** join/#asterisk shay|work (i=shay@unaffiliated/shay) |
| 08:02.18 | shay|work | hi |
| 08:02.47 | *** join/#asterisk chris_asterisk` (n=test@217.205.167.170) |
| 08:02.52 | chris_asterisk` | <PROTECTED> |
| 08:03.12 | Gido-E | kphone |
| 08:03.27 | chris_asterisk` | I'll take a look, cheers Gido-E :) |
| 08:03.44 | chris_asterisk` | Is that Winblows based? |
| 08:04.02 | Gido-E | chris_asterisk` do you think i am santaclaus? |
| 08:04.19 | fetcher | do any of the free SIP softphones support video? |
| 08:04.30 | chris_asterisk` | haha, I should have guessed, I shoulda been more specific :P |
| 08:05.44 | chris_asterisk` | xTen does video i think fetcher, not sure if its free. doubt it. |
| 08:08.59 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 08:14.39 | *** join/#asterisk gr1ncheux (n=devine@unaffiliated/gr1ncheux) |
| 08:20.20 | *** join/#asterisk Ebola (n=Ebola@host86-136-130-202.range86-136.btcentralplus.com) |
| 08:28.29 | *** join/#asterisk friedrich| (n=friedric@e177255054.adsl.alicedsl.de) |
| 08:40.59 | hermuli | anyone using varphonex with asterisk? |
| 08:41.24 | *** join/#asterisk Chris-NB (n=chris@ip.tech.t-mobile.at) |
| 08:41.50 | hermuli | i'm just getting "there's nobody available to take your call" when i try to call to my asterisk and there's not even a blink in CLI and i'm just wondering what am i not getting here (again) |
| 08:43.50 | *** join/#asterisk jm|work (n=jm@sentry.flags.co.uk) |
| 08:45.54 | *** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25) |
| 08:46.32 | Chris-NB | hermuli, are you placing a sip call? |
| 08:47.51 | hermuli | yes |
| 08:50.37 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 08:50.51 | *** join/#asterisk mquin (n=mike@pdpc/supporter/active/mquin) |
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| 08:51.52 | *** join/#asterisk `p4r14h`work (n=josh@24-119-48-78.cpe.cableone.net) |
| 08:52.27 | hermuli | need to reboot... |
| 08:54.55 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 08:55.54 | *** join/#asterisk ValDuane (n=valduane@cpc4-basf6-0-0-cust470.nott.cable.ntl.com) |
| 08:56.55 | ValDuane | I have recently bought a TE205P and can't seem to get the drivers installed anyone who can help? |
| 09:00.50 | *** join/#asterisk Fieldy (i=qxk7gJM7@gentoo/contributor/Fieldy) |
| 09:06.27 | *** join/#asterisk hermuli (n=Eladamri@a88-112-255-26.elisa-laajakaista.fi) |
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| 09:18.13 | Chris-NB | hermuli, enable sip debug and place your call again |
| 09:19.16 | Chris-NB | hermuli, probably no extension matches your incoming call. look at sip debuging and see what nr is dialed (sip calls come in in default context) |
| 09:19.32 | hermuli | ok, thanks. i'll try |
| 09:19.49 | Chris-NB | k |
| 09:24.44 | *** join/#asterisk shinux__ (n=shinux@196.201.157.211) |
| 09:27.37 | *** join/#asterisk ivanfm (n=ivanfm@c93481ec.virtua.com.br) |
| 09:28.46 | *** join/#asterisk chris_asterisk` (n=test@217.205.167.170) |
| 09:29.01 | chris_asterisk` | hey guys, does anyone know if Asterisk 1.2 supports SIP call transfering, preferably via SJPhone? |
| 09:31.48 | *** join/#asterisk shinux__ (n=shinux@196.201.157.211) |
| 09:44.43 | *** join/#asterisk Azoff (i=foobar@host106-7.junet.se) |
| 09:44.45 | Azoff | hello |
| 09:52.22 | *** join/#asterisk ivanfm_ (n=ivanfm@c93481ec.virtua.com.br) |
| 09:52.59 | *** join/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 09:53.48 | nextime | Hi. Is there any reference on all options that i can put on users.conf or i really need to start reading the code of all channels that support it? |
| 10:07.56 | mvanbaak | anyone here using voop ? |
| 10:10.26 | nextime | what's voop? |
| 10:13.22 | mvanbaak | voop is an itsp |
| 10:16.41 | *** part/#asterisk oej (n=olle@apollo.webway.se) |
| 10:32.19 | *** join/#asterisk Chris-NB (n=chris@ng1.kurtkrenn.com) |
| 10:35.42 | hermuli | Chris: thank you i got it to work |
| 10:35.45 | *** join/#asterisk nitram (i=foo@superblob.com) |
| 10:42.23 | Chris-NB | hermuli, great. |
| 10:46.31 | *** join/#asterisk MrWup (n=neil@80-46-38-124.static.dsl.as9105.com) |
| 10:46.33 | MrWup | hey guys |
| 10:46.45 | MrWup | how do increase the overall volume of playbacked wav files? |
| 10:46.51 | MrWup | asterisk seems to be playing them too quietly |
| 10:50.53 | *** join/#asterisk abuyazan (n=khaled@dogbert.palnet.com) |
| 10:53.55 | tzafrir | MrWup, change the sound files :-( |
| 10:54.18 | tzafrir | MrWup, to which phone exactly? |
| 10:54.56 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/active/cpm) |
| 10:55.33 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 10:57.13 | *** join/#asterisk hot_wheelz (i=hotwheel@203-214-12-140.dyn.iinet.net.au) |
| 10:59.02 | *** join/#asterisk reber (n=reber@cl-158.dub-01.ie.sixxs.net) |
| 11:00.00 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 11:01.31 | joobie | guys anyone know where i can get polycom phones in australia (preferably melbourne) |
| 11:01.43 | joobie | i found a site but they are like 3x the price of US sites online |
| 11:06.55 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 11:07.32 | MrWup | urgh the playback quality if awful |
| 11:07.36 | *** join/#asterisk Tclp (n=tcalp@S01060016b680c0e0.ed.shawcable.net) |
| 11:07.41 | MrWup | my wav file sounds great when played through my headphones |
| 11:07.49 | MrWup | voice quality between sip handsets also sounds great |
| 11:07.57 | MrWup | but converting the wav to gsm and then playing it sounds awful |
| 11:09.05 | Tclp | hey all .. I'm using TFOT book as a reference for installing asterisk (first time n00b) ... it's saying I should download / install asterisk sounds package .. I don't see it in the asterisk directory on digium ? ... has it since been included into the main package or ? |
| 11:22.34 | Dr-Linux | can i use sip client info in my asterisk server? |
| 11:24.40 | Dr-Linux | as a trunk? |
| 11:28.28 | zeeesh | at the same server p2p is simple, if i have 2 peer and both are at different machines then how can connect with each other???? |
| 11:28.28 | Dr-Linux | anybody active :P |
| 11:32.39 | *** join/#asterisk Bhaal (i=bhaal@freenode/staff/bhaal) |
| 11:34.26 | MrWup | anyone know why sound quality is so bad |
| 11:34.37 | MrWup | playback with gsm really sounds bad on my phones |
| 11:34.47 | MrWup | which are capable of playing damngood sound quality |
| 11:34.59 | MrWup | is there any way asterisk can raise the quality of its playback? |
| 11:36.05 | giasai68 | hello |
| 11:36.29 | giasai68 | anyone able to send me a fax t38 passthrough sample config? |
| 11:37.22 | giasai68 | anyone there? |
| 11:39.16 | *** join/#asterisk chris_asterisk` (n=test@217.205.167.170) |
| 11:39.25 | chris_asterisk` | hey guys, probably a really quick easy question. |
| 11:39.26 | chris_asterisk` | If i want CDR information to write to remote mySQL database, is it only cdr_mysql.conf that needs to be altered? |
| 11:42.35 | *** join/#asterisk nasls_lsa (n=chatzill@athedsl-140503.home.otenet.gr) |
| 11:44.25 | *** join/#asterisk friedrich| (n=friedric@e177246147.adsl.alicedsl.de) |
| 11:45.56 | chris_asterisk` | :o |
| 11:48.36 | *** join/#asterisk zike (n=z1@212.248.177.114) |
| 11:50.58 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 11:51.35 | GreyFoxx | Chris-NB: well, cdr_mysql.conf and of course the remote mysql machine has to be configured to allow the sql user to connect from your * machine |
| 11:51.46 | GreyFoxx | err that should have been chris_asterisk |
| 11:52.05 | Chris-NB | np : D |
| 11:53.42 | chris_asterisk` | GreyFoxx : i've changed the cdr_mysql.conf (the db server allows connections from * currently), if i go to call reports it's full of CDR data.. |
| 11:53.53 | chris_asterisk` | the new table doesn't have any cdr data, so it seems like it's still reading from localhost. |
| 11:54.36 | *** join/#asterisk hads (n=hads@reef80.anchor.net.au) |
| 11:55.02 | chris_asterisk` | box has been restarted too. |
| 11:56.22 | tzanger | woo |
| 11:56.34 | chris_asterisk` | yea, tis, when i stop mysqld it throws a hissy fit. |
| 11:57.09 | chris_asterisk` | res_mysql.conf is for all other mySQL/asterisk stuff, so i can't think of anywhere else to look. |
| 11:57.42 | GreyFoxx | Chris-NB: pastebin your cdr_mysql.conf contents |
| 11:57.48 | GreyFoxx | damn auto complete |
| 11:57.55 | Chris-NB | *hehe |
| 11:58.29 | chris_asterisk` | Righto. |
| 11:59.01 | chris_asterisk` | http://pastebin.ca/423749 |
| 11:59.23 | GreyFoxx | comment out the sock line |
| 11:59.45 | chris_asterisk` | if that config fails does it by default load the localhost? |
| 12:00.08 | GreyFoxx | chris_asterisk`: never tested that |
| 12:00.23 | chris_asterisk` | I disabled mysql on localhost, commented out the sock line. |
| 12:00.26 | chris_asterisk` | and get : nativecode=Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2)] ** mysql://asteriskuser:amp109@localhost/asterisk |
| 12:00.37 | chris_asterisk` | the default connection string heh. |
| 12:01.04 | chris_asterisk` | this is via reports on freepbx, thinking about it, would they have their own mysql configuration files? |
| 12:01.35 | GreyFoxx | no clue, never looked at freepbx, but it sounds like it's not using that config file for the mysql info :) |
| 12:01.59 | chris_asterisk` | indeedily :) |
| 12:02.00 | chris_asterisk` | time to dig around. |
| 12:02.45 | *** join/#asterisk ikaRus1 (i=none@80.179.36.48.static.012.net.il) |
| 12:03.05 | GreyFoxx | freepbx might require cdr are on the same box. *shrug* |
| 12:03.20 | chris_asterisk` | nah, have seen it elsewhere on a previous installation i believe, just not sure how it was done :| joy! |
| 12:03.47 | ikaRus1 | is there a way to specify calls-limit globaly instead of having to insert it to all the sip peers manually? |
| 12:03.57 | *** join/#asterisk Ahrimanes (n=ma@81.7.159.2) |
| 12:11.08 | Tclp | can someone pls help me with gettings asterisk installed ? |
| 12:11.24 | Tclp | It keeps telling me that I don't have the linux kernal source installed ... |
| 12:11.43 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:12.07 | e-ddie | it might be an idea to install it |
| 12:12.15 | e-ddie | if it tells you you havent |
| 12:12.25 | Tclp | I'm running kubuntu 6.06, I've downloaded the closest match to the kernal that I could and even created a symbolic link to the kernal revision |
| 12:13.55 | Tclp | is there a linux distrobution that has asterisk / zaptel / etc .. pre-installed ? :) |
| 12:14.49 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 12:15.40 | ikaRus1 | Tclp:: try this: https://wiki.ubuntu.com/AsteriskOnUbuntu |
| 12:16.34 | Tclp | hrm |
| 12:16.35 | ikaRus1 | Tclp:: ot this: http://www.aussievoip.com/wiki/freePBX-Ubuntu |
| 12:16.45 | Tclp | is there any disadvantages to running tribox ? |
| 12:16.55 | Tclp | sounds a lot easier :) |
| 12:16.57 | ikaRus1 | ummm |
| 12:17.31 | ikaRus1 | dunno, i like to do things myway, asterisk/freepbx, latest versions |
| 12:18.26 | ikaRus1 | there are ready made packages for debian/ubuntu |
| 12:20.10 | *** join/#asterisk friedrich| (n=friedric@e177246147.adsl.alicedsl.de) |
| 12:26.29 | Tclp | I think Tribox may be my best bet |
| 12:26.33 | Tclp | seems easy |
| 12:26.44 | Tclp | being that I've never run linux/unix before I'm pretty lost as it is |
| 12:26.58 | Tclp | nor am I any sort of pro when it comes to telephony |
| 12:28.04 | ikaRus1 | go for it then.... |
| 12:28.04 | ikaRus1 | gotta go |
| 12:28.06 | *** part/#asterisk ikaRus1 (i=none@80.179.36.48.static.012.net.il) |
| 12:29.31 | Tclp | man fast internet is great (d/l'ing @ 1.8MB/s) |
| 12:37.11 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 12:37.56 | *** join/#asterisk [Airwolf] (n=airwolf@voip.ymav.nl) |
| 12:39.02 | ghenry | anyone tried the Aastra 55i IP Telephone and 53i? |
| 12:40.36 | *** join/#asterisk ming_zym (n=ming_zym@124.254.56.42) |
| 12:42.00 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 12:42.34 | *** join/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 12:44.05 | Dr-Linux | can i use sip client info in my asterisk server? |
| 12:44.09 | Dr-Linux | as a trunk? |
| 12:44.20 | *** join/#asterisk zeeesh (i=GrTB0T@202.38.55.125) |
| 12:44.21 | zeeesh | hi |
| 12:48.22 | zeeesh | p2p at the same server its easy .. i need to know if i have 2 peers at 2 different machines then how can i make a call from p1 to p2 ??? |
| 12:48.48 | blitzrage | zeeesh: DUNDi |
| 12:49.20 | blitzrage | DUNDi + regcontext + regexten == distributed peers |
| 12:52.25 | *** join/#asterisk mmartinn (n=martins@128.227.123.22) |
| 12:55.53 | *** join/#asterisk Meaty` (n=meaty3@office.abi.ca) |
| 12:56.12 | MrWup | is there any way of signalling to the user their status |
| 12:56.12 | MrWup | like |
| 12:56.18 | *** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn) |
| 12:56.27 | blitzrage | MrWup: you mean presence? |
| 12:56.31 | MrWup | say if the person is in DND mode, can they jsut lift the handset and it will say, "you are in DND" mode? |
| 12:56.37 | MrWup | i.e. without them having to dial something |
| 12:56.42 | blitzrage | MrWup: not with a SIP phone |
| 12:56.51 | blitzrage | possible on Zap channels |
| 12:57.00 | blitzrage | *analogue |
| 12:57.21 | MrWup | cause basically i want a user to be able to hit a programmed button and go into a mode. then when someone calls the person it says "x is in dnd mode, to disturb, press 1, to cancel, press 2" |
| 12:57.36 | blitzrage | because the SIP phone doesn't send signalling until someone dials something |
| 12:57.52 | MrWup | and just so the person who is in dnd mode does not forget, i need some way of alerting the user theyre in the mode |
| 12:57.53 | blitzrage | MrWup: that case you can program in the dialplan |
| 12:59.29 | MrWup | like a global variable or something? |
| 12:59.32 | MrWup | cause i want two modes |
| 12:59.54 | MrWup | e.g. DND mode, and also "out of office" mode, so the person can hit a button before they leave and the receptionist will know their status |
| 13:00.13 | mmartinn | Is there *any* reason ${EXTEN} would contain callerid instead of a number? ChanSpy is crashing my asterisk and shows itself attaching to my outgoing caller id instead of the channel. It only happens rarely. |
| 13:01.13 | blitzrage | MrWup: yes -- that is all dialplan programming -- as long as the user dials something, you can make Asterisk do what you want |
| 13:01.22 | MrWup | ok |
| 13:01.29 | blitzrage | MrWup: the way to know status is via presence |
| 13:01.49 | MrWup | in asterisk how do i catch when a user dials *50 |
| 13:01.58 | MrWup | cause i want three numbers to always be phone extensions |
| 13:02.07 | MrWup | and * and two numbers to be like special functions |
| 13:03.30 | kaldemar | exten => *50,1,dosomething |
| 13:03.39 | MrWup | mm thats what i thought |
| 13:03.50 | MrWup | but i press * and my phone stops accepting keystrokes |
| 13:04.07 | blitzrage | mmartinn: yah -- you have a bug in your dialplan somewhere that is doing something like Set(CALLERID(number)=${EXTEN}) |
| 13:04.16 | kaldemar | do you have anything starting with * in your dialplan that the particular phone can see? |
| 13:04.20 | blitzrage | MrWup: that's because of the phone's dialplan |
| 13:04.28 | MrWup | oh |
| 13:04.34 | blitzrage | Polycom? |
| 13:04.38 | MrWup | aastra |
| 13:04.40 | blitzrage | or Linksys SPA? |
| 13:04.46 | blitzrage | ahhh -- I don't know anything about the Aastra's |
| 13:04.53 | blitzrage | but that's why that happens |
| 13:04.54 | MrWup | but then i changed the dialplan to take three numbers |
| 13:04.57 | MrWup | my fault i supose |
| 13:05.07 | blitzrage | you need to tell the phone to stop trying to parse on * immediately |
| 13:05.08 | mmartinn | blitzrage: Wouldn't that just set caller id and not affect ${EXTEN}? |
| 13:05.14 | MrWup | yeah |
| 13:05.18 | MrWup | hmm |
| 13:05.20 | MrWup | presence ey |
| 13:05.23 | mmartinn | blitzrage: I don't have Set(${EXTEN... anywhere |
| 13:05.31 | MrWup | so with presence i can do that whole case i described? |
| 13:05.38 | blitzrage | mmartinn: I mis-read -- I have no idea what that'd happen :) |
| 13:05.51 | blitzrage | MrWup: go read about presence and come to your own conclusions |
| 13:06.05 | MrWup | the wiki has 0 on presence |
| 13:06.18 | blitzrage | incorrect |
| 13:06.23 | blitzrage | look for 'hints' |
| 13:07.06 | blitzrage | http://voip-info.org/wiki/view/Asterisk+presence |
| 13:07.13 | blitzrage | or how about "Asterisk and presence" |
| 13:08.41 | MrWup | mm thats what im reading |
| 13:08.50 | MrWup | not much to read |
| 13:08.52 | MrWup | very vague |
| 13:08.58 | MrWup | obviously simple if u grasp the concept |
| 13:10.05 | MrWup | that just links SIP users with extensions right? |
| 13:11.16 | blitzrage | it links SIP users with a hint |
| 13:11.58 | MrWup | could you give me a practical example? this all doesn't mean anything to me |
| 13:12.22 | blitzrage | exten => username,hint,SIP/username |
| 13:12.31 | MrWup | yeah ive seen that |
| 13:12.35 | MrWup | but what does it actually do? |
| 13:12.46 | blitzrage | phones SUBSCRIBE to the hint, then update their status with Asterisk |
| 13:12.58 | MrWup | what does that mean? |
| 13:13.46 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:14.07 | blitzrage | MrWup: SUBSCRIBE is a type of message a SIP phone uses to monitor the status of things |
| 13:18.30 | mmartinn | blitzrage: do you think ChanSpy could have a bug that would say it was attaching to the CallerID field instead of a channel name? Perhaps right as a call is being built/bridged? |
| 13:19.09 | blitzrage | mmartinn: not in my experience -- but anything is possible -- you have to be able to isolate the issue though or the developers aren't going to know where to look |
| 13:19.40 | mmartinn | blitzrage: We're suspecting a bug or a bad ATA, but I can't fathom how an ATA would maybe ask ChanSpy to do anything like this |
| 13:24.56 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:24.56 | *** mode/#asterisk [+o mog] by ChanServ |
| 13:31.18 | *** join/#asterisk uwe (n=uwe@dogbert.palnet.com) |
| 13:34.57 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 13:34.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:35.05 | uwe | um, hello, if you had a strange behaviour on some extensions in a queue with Ringall strategy (dont recive calls.. or rarely recive calls), and you need to invistigate it, what would you do?? i did check dnd (database show dnd), i do sip show peers and ping the peers, i call extensions directly ... what else can be checked ? |
| 13:35.58 | uwe | given that this behaviour is not easily reproducable ... it comes after asterisk has been running for tens of hours. |
| 13:36.11 | uwe | like 20 hours |
| 13:36.23 | giasai68 | hello, i'm tring to send a fax with capanga software using asteris as pbx and terminating on a ATA box connected to a analogic fax (using t.38). tha connection of fax work, when the fax start to send page nr. 1 after 40 secs i get fax failed tx. in asteris i enable debug4 verbosity 4 and sip debug and i get this:http://pastebin.ca/423872 any saggestions? |
| 13:36.56 | *** part/#asterisk nextime (n=nextime@unaffiliated/nextime) |
| 13:39.28 | SplasPood | Anyone know of a nice set of docs for the Voicemail(Main) system that would be useful to end users? |
| 13:46.04 | uwe | SplasPood, um, what about just giving them the voicemail feature codes |
| 13:47.03 | SplasPood | uwe: what do you mean? |
| 13:47.53 | uwe | what would be useful for the end users ? ... how to use voicemail i suppose |
| 13:47.58 | SplasPood | I realize that the voicemail system is rather self-explanatory, but my clients are... retards |
| 13:48.45 | uwe | oh ... |
| 13:48.47 | uwe | LOL |
| 13:49.21 | blitzrage | SplasPood: there aren't that many -- just go through them and write some custom docs |
| 13:49.25 | *** join/#asterisk malverian (n=malveria@gentoo/developer/malverian) |
| 13:49.50 | uwe | so you need a small article about what voice mail is ... |
| 13:49.53 | SplasPood | blitzrage: yea.. thats the route I may have to go... I seem to recall seeing a PDF once with a full tree of the options at some point |
| 13:49.59 | uwe | try wikipedia.org SplasPood |
| 13:50.03 | SplasPood | uwe: not what it is.. but.."how do I set my greeting?" |
| 13:50.49 | *** join/#asterisk Nivex (n=kjotte@user-0ce2kma.cable.mindspring.com) |
| 13:50.54 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 13:53.59 | blitzrage | SplasPood: "very carefully" |
| 13:54.29 | SplasPood | blitzrage: heh totally. |
| 13:54.37 | blitzrage | ok... back to work for me after breakfast |
| 13:54.38 | *** part/#asterisk blitzrage (n=blitzrag@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:57.23 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 13:57.23 | *** mode/#asterisk [+o mog] by ChanServ |
| 14:00.03 | *** join/#asterisk Fieldy (i=KTT7DtYl@gentoo/contributor/Fieldy) |
| 14:07.32 | *** join/#asterisk neoalex (n=neoalex@user-0ccenic.cable.mindspring.com) |
| 14:08.58 | neoalex | hi guys, I have a weird problem with my asterisk |
| 14:09.24 | neoalex | I have set up my extensions so that I can call my server, and get an ivr to call abroad from there |
| 14:09.52 | chris_asterisk` | Anyone use ARI? i've got asterisk storing CDR data in an external mySQL db, how do I get the ARI and reports.php to read from that? |
| 14:10.19 | neoalex | it all work fine... the only problem is that, when I call abroad, I only get one short ring |
| 14:10.45 | *** join/#asterisk MedozasSVR (n=MedozasS@p54911513.dip0.t-ipconnect.de) |
| 14:10.54 | neoalex | during the remote phone actually rings |
| 14:10.59 | neoalex | I just can't hear it |
| 14:11.01 | *** join/#asterisk MrWup (n=neil@80-46-38-124.static.dsl.as9105.com) |
| 14:11.11 | MedozasSVR | Hi guys - does anybody have an idea to FIND as substring within a variable in asterisk?' |
| 14:11.51 | MedozasSVR | example: string "sip:348729487294@test.net" and i want to find 348729487294 and return true or false, is that possible? |
| 14:12.21 | MedozasSVR | any help would be greatly appreciated |
| 14:13.35 | *** join/#asterisk jcmoore (n=jcmoore@unaffiliated/tgrman) |
| 14:16.16 | *** join/#asterisk putzz (n=me@CPE00155824933c-CM00111ae07ac2.cpe.net.cable.rogers.com) |
| 14:18.43 | *** join/#asterisk fo0b (n=rick@81-86-30-78.dsl.pipex.com) |
| 14:19.41 | kumbalae | MedozasSVR: why don't you take it to agi and do all the processing ? |
| 14:19.48 | *** join/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 14:19.55 | MedozasSVR | ara (realtime) |
| 14:20.19 | fo0b | Is there a way to get asterisk 1.2.1 to set the permissions on voicemail files to something a little less restrictive? I've tried changing the umask for my asterisk user, yet it only effects the permissions of the msgXXXX.txt files, not the wav's. |
| 14:20.23 | kumbalae | whether realtime or non realtime, that is ok |
| 14:20.34 | kumbalae | agi works with your requirement in few lines |
| 14:20.52 | MedozasSVR | well, multiple asterisks, all database based, no flat files (all db), no agi, nothing |
| 14:21.01 | MedozasSVR | even voicemail system in db via blob |
| 14:21.37 | kumbalae | what db you use ? |
| 14:21.45 | MedozasSVR | mysql |
| 14:21.55 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 14:21.59 | MedozasSVR | actually best for this purpose |
| 14:22.15 | kumbalae | database is common to all asterisk ? |
| 14:22.27 | MedozasSVR | i dont understand what you mean? |
| 14:23.31 | MedozasSVR | well, you can use the ara (asterisk realtime) with any db which you can access with odbc, but mysql is the most native solution, as there is direct support in asterisk-addons |
| 14:24.35 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 14:24.43 | MedozasSVR | and actually as we've set up a HA (high availability) asterisk cluster, we don't use any flat files. everything is stored in db and replicated via mysl-cluster engine, so there is no single-point-of-failure |
| 14:25.16 | MedozasSVR | but i think ive found my solution already - will post it in few min |
| 14:26.15 | Dr-Linux | can i use sip client info in my asterisk server? |
| 14:26.17 | Dr-Linux | as a trunk? |
| 14:26.34 | kumbalae | Dr-Linux: yes |
| 14:27.32 | *** join/#asterisk ManxPower (n=manxpowe@191.sub-75-202-34.myvzw.com) |
| 14:27.37 | Dr-Linux | kumbalae: like: |
| 14:27.38 | Dr-Linux | if i was given, i sip username info |
| 14:27.44 | Dr-Linux | username, password, sipserver |
| 14:27.51 | kumbalae | use sip realtime |
| 14:27.52 | Dr-Linux | and i'm using it just fine with a sip softphone |
| 14:27.57 | Dr-Linux | but now i wanna use the same sip account in my asterisk server |
| 14:27.59 | kumbalae | and store it in the mysql |
| 14:28.18 | kumbalae | Dr-Linux: Asterisk Realtime is the solution for that |
| 14:28.35 | Dr-Linux | kumbalae: actually i know how to use sip trunk between two servers |
| 14:28.40 | Dr-Linux | but here is little different case |
| 14:28.42 | ManxPower | asterisk realtime is seldom a solution. |
| 14:28.48 | kumbalae | http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip |
| 14:29.02 | Dr-Linux | kumbalae: i know how to use ARA |
| 14:29.08 | Dr-Linux | but my question is something else |
| 14:29.18 | ManxPower | Dr-Linux: I came in late in the conversation. Can you give me a summary of your question? |
| 14:29.18 | Dr-Linux | i don't have access on my sip provider server |
| 14:29.39 | *** join/#asterisk mbranca (n=matteo@host-210-mi.linuxserver.it) |
| 14:29.41 | Dr-Linux | ManxPower: my question looks very stupid maybe, but lemme ask |
| 14:29.49 | Dr-Linux | can i use sip client info in my asterisk server? |
| 14:29.53 | Dr-Linux | as a trunk? |
| 14:30.04 | Dr-Linux | if i was given, i sip username info |
| 14:30.09 | Dr-Linux | username, password, sipserver |
| 14:30.37 | ManxPower | Dr-Linux: There is no such thing as a "sip trunk" in Asterisk. But you can set Asterisk up as a SIP client to the provider. |
| 14:30.38 | Dr-Linux | and i'm using it on sip softphone which connects to their server |
| 14:31.03 | Dr-Linux | but i wanna use the same account on the server. |
| 14:31.10 | ManxPower | It would depend on the provider if they allow you to run two SIP clients against the same account, of course. |
| 14:31.18 | neoalex | when I forward place a call through an ivr menu (meaning when I press one it dials SIP/123456@somepeer) I can't hear the ring |
| 14:31.31 | neoalex | I only hear a short ring |
| 14:31.40 | ManxPower | neoalex: you do not have /etc/asterisk/indications.conf |
| 14:31.42 | Dr-Linux | ManxPower: they allow me 10 calls at same time |
| 14:31.57 | ManxPower | You hear no ring or a short ring, neoalex? |
| 14:32.05 | neoalex | a short ring' |
| 14:32.14 | neoalex | like half of ring |
| 14:32.15 | Dr-Linux | ManxPower: so any hint doing that on the server? i mean in sip.conf |
| 14:32.17 | ManxPower | Dr-Linux: just create an entry in sip.conf using that user/pass/hostname |
| 14:32.23 | neoalex | even though the remote phone is ringing fine |
| 14:32.30 | *** join/#asterisk Dovid (n=Dovid@l192-117-114-1.broadband.actcom.net.il) |
| 14:32.30 | Dr-Linux | ManxPower: i'm not sure if i'll do the same as i do for my normal sip clients |
| 14:32.42 | neoalex | and I do have indications.conf (never opened it though) |
| 14:32.46 | *** join/#asterisk `p4r14h (n=j0sh@69.92.145.178) |
| 14:32.52 | neoalex | it's just the one that comes with the samples |
| 14:32.53 | ManxPower | neoalex: indications.conf tells asterisk how to provide ring sounds AFTER a call has been answered, like in the case of a person using an IVR. |
| 14:33.04 | Dr-Linux | ManxPower: okey? |
| 14:33.11 | ManxPower | Dr-Linux: Give me a min, I'll whip up an example. |
| 14:33.21 | Dr-Linux | ManxPower: thanks |
| 14:33.31 | ManxPower | Dr-Linux: do you only make calls via that provider or do you make and receive calls via that provider? |
| 14:33.51 | Dr-Linux | ManxPower: only outgoing through provider |
| 14:34.50 | neoalex | ok... well in indications.conf my default location is setup correctly |
| 14:35.01 | neoalex | other then that... I don't know what I should change |
| 14:37.53 | ManxPower | Dr-Linux: everything surrounded by ** needs to be changed to suit your account |
| 14:38.36 | Dr-Linux | ManxPower: ok thanks |
| 14:38.52 | ManxPower | neoalex: not really. pastebin the CLI output of a call from the time you select the IVR option to the time the call hits voicemail. Make sure the person does not answer. |
| 14:39.53 | *** join/#asterisk _deg_ (n=deg@200.195.161.164) |
| 14:40.02 | _deg_ | where should I look to change de transfer callerid behaviour? app_dial.c ? |
| 14:40.15 | _deg_ | I mean. I need to transfer a call using my callerID as the A number instead of the callers number.... |
| 14:40.28 | ManxPower | _deg_: what SIP device are you using? |
| 14:40.36 | _deg_ | Polycom |
| 14:40.42 | _deg_ | Is one of them... |
| 14:40.49 | _deg_ | I have some others... |
| 14:40.57 | ManxPower | _deg_: you do this using the Blind Transfers |
| 14:41.01 | _deg_ | Is this a UA problem? |
| 14:41.16 | _deg_ | blind and attendant |
| 14:41.19 | _deg_ | both |
| 14:41.21 | ManxPower | On Polycom it would be during a call TRANSFER then BLIND |
| 14:41.25 | ManxPower | then the number |
| 14:41.50 | *** join/#asterisk mathai (n=root@dvere.psg.sk) |
| 14:41.55 | _deg_ | Im using my own features.conf numbers to perform the transfers. |
| 14:42.10 | ManxPower | _deg_: by default Dial uses the original CLID in a blind transfer, since a supervised transfer is really two calls that are conferenced in, I don't know if you can get the original CLID. |
| 14:42.40 | ManxPower | _deg_: then use blind transfers |
| 14:42.41 | Dr-Linux | ManxPower: where is the example? :S |
| 14:42.47 | ManxPower | http://pastebin.ca/423964 |
| 14:42.53 | ManxPower | Dr-Linux: sorry |
| 14:42.59 | Dr-Linux | im sorry |
| 14:43.00 | neoalex | ManxPower: http://www.pastebin.us/21190 |
| 14:43.01 | ManxPower | It helps when you actually PASTE the URL. |
| 14:43.03 | Dr-Linux | i think you were waiting for me :P |
| 14:43.08 | *** join/#asterisk genz (n=chatzill@im.jobdig.com) |
| 14:43.18 | ManxPower | Well, when I actually paste the URL. |
| 14:43.57 | neoalex | brb... |
| 14:44.04 | Dr-Linux | ManxPower: opss, just that? :S |
| 14:44.11 | Dr-Linux | aww |
| 14:44.15 | ManxPower | Dr-Linux: what you want is incredibly simple |
| 14:44.21 | mathai | Hi, does anybody know why delivered Xorcom TS-1 can't detect Astribank 16? |
| 14:44.24 | _deg_ | ManxPower, The problem is that de Session serer fdrom Nortel will not allow calls with a callerid that is not reconized as accounts of its database... When tranfer a call from outside to an outside number the Sesssion server will receive an external callerID and will drop de call... |
| 14:44.38 | Dr-Linux | ManxPower: actually i never know about other sip providers, |
| 14:44.54 | Dr-Linux | so i thought that would be some different seience :P |
| 14:44.57 | ManxPower | _deg_: I don't really have any suggestions for a solution to that problem. |
| 14:45.23 | _deg_ | ManxPower, thanks anyway. |
| 14:45.25 | ManxPower | Dr-Linux: As far as Asterisk is concerned a SIP provider is just a SIP phone that can accept MANY calls at the same time. |
| 14:45.34 | _deg_ | ManxPower, maybe the source code should be my solution.... |
| 14:45.47 | ManxPower | ~mailinglist |
| 14:45.48 | jbot | Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 14:45.54 | *** join/#asterisk JoeDeveloper (n=jdevel@www.airlinksystems.com) |
| 14:45.55 | Dr-Linux | ManxPower: thank you |
| 14:45.59 | chris_asterisk` | sod it, trixbox is just bloatware |
| 14:46.00 | ManxPower | _deg_: search the mailing list archives. |
| 14:46.11 | _deg_ | _deg_, I will do that. |
| 14:46.13 | ManxPower | chris_asterisk`: it has its place, but not here. |
| 14:46.40 | _deg_ | ManxPower, Do you have any american keyword that I should use? Im not a native english speaker. |
| 14:46.42 | ManxPower | _deg_: the Nortel should be able to be configured to accept any CLID |
| 14:47.11 | JoeDeveloper | How do I check the return value from exten => s,n,SpeechLoadGrammar(grammar1|grammar1_file)? |
| 14:47.23 | ManxPower | _deg_: Perhaps: Nortel SIP transfer |
| 14:47.33 | _deg_ | ManxPower, Yes, but on this customer thay have some particular needs... |
| 14:47.33 | ManxPower | _deg_: The actual product name would be a good start |
| 14:47.38 | _deg_ | Ok. |
| 14:47.55 | _deg_ | The old one was called MCS something... I will check the new one. |
| 14:48.11 | ManxPower | JoeDeveloper: as the exten => s,n,Noop(${THE VARIABLEYOUWANT}) |
| 14:48.43 | ManxPower | JoeDeveloper: Noop just does nothing but print the value of the variables on the CLI |
| 14:49.00 | *** join/#asterisk marv[work] (n=timr@24.214.206.254) |
| 14:49.27 | genz | Is there a setting that controls the recording volume? Some of our voicemail recordings are very loud. |
| 14:49.46 | *** join/#asterisk badcfe (n=cso@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 14:50.22 | badcfe | is there some way of monitoring rfc2833 dtmf as they arrive by some other means than the * cli ? |
| 14:50.25 | JoeDeveloper | ManxPower: what are you talking about? are you saying there is no way to check if the grammar is loaded by the speechloadgrammar command? |
| 14:50.31 | neoalex | ManxPower here's the CLI output you requested : http://www.pastebin.us/21190 |
| 14:51.07 | genz | Anyone relying on cdr_mysql and using 1.4? |
| 14:51.15 | fo0b | Is there a way to get asterisk 1.2.1 to set the permissions on voicemail wav files?? umasking only changes the .txt files created in /var/spool/asterisk/voicemail |
| 14:51.50 | badcfe | i have dtmfmode=rfc2833 in the sip peer but cant see dtmf arrive. the sender uses rfc2833 but i want to double check that without * |
| 14:51.52 | MrWup | anyone here use the aastra 9133i? |
| 14:52.42 | MrWup | im having trouble finding out how to use the BLT and presence stuff |
| 14:52.59 | MrWup | so i can do like out to lunch, in a meeting, seeing client etc custom states |
| 14:53.19 | ManxPower | neoalex: I think your provider is answering the phone before processing the call. I cannot think of any other reason for what you are experiencing. |
| 14:53.46 | ManxPower | JoeDeveloper: No. They only way to tell is of the application sets a dialplan variable and then you would check the dialplan variable. |
| 14:54.14 | neoalex | ManxPower: I'm experiencing the same with two different providers |
| 14:54.50 | ManxPower | neoalex: You can try removing the "r" from the Dial line. That option usually causes problems. |
| 14:55.05 | ManxPower | neoalex: other than that I don't have any more suggestions. |
| 14:55.45 | *** join/#asterisk PMantis (n=pmantis@cpe-74-65-53-178.rochester.res.rr.com) |
| 14:56.48 | ManxPower | JoeDeveloper: do a "show application speechloadgrammar" and put the output on pastebin.ca |
| 14:57.05 | fo0b | I've found this bug http://bugs.digium.com/view.php?id=6334 relating to the issue I'm talking about, but I'm not sure what version of asterisk it applies to? |
| 14:57.09 | ManxPower | since I don't have that command I can't help you unless I see the docs for that command. |
| 14:57.15 | PMantis | I think Asterisk 1.4.2 has a bug. After setting a digit timeout, and "Background"ing a file, * just hangs up on people, rather than waiting for the timeout, and jumping to the 't' exten. |
| 14:57.46 | genz | PMantis: Talking about IVR stuff? |
| 14:57.52 | PMantis | Ye |
| 14:57.56 | PMantis | Yes |
| 14:57.57 | JoeDeveloper | Does anyone know what variable the SpeechLoadGrammar() command stores its return value in (or any of the SpeechXXX commands store their return vars) |
| 14:58.06 | ManxPower | fo0b: are you running SVN trunk or released version of Asterisk |
| 14:58.11 | *** join/#asterisk KingCoco (n=KingCoco@89.106.200.61) |
| 14:58.18 | JoeDeveloper | ManxPower: where do you want me to run "show application speechloadgrammar"? |
| 14:58.24 | ManxPower | JoeDeveloper: "show application X" will give you the docs for the X application |
| 14:58.29 | ManxPower | JoeDeveloper: in the Asterisk CLI |
| 14:58.31 | fo0b | ManxPower, a 1.2.1 release |
| 14:58.41 | ManxPower | fo0b: then it does not apply to your release. |
| 14:59.00 | ManxPower | fo0b: SVN trunk is never a release version |
| 14:59.02 | neoalex | ManxPower: removing ring I can't hear any r, are there any other options I could use |
| 14:59.06 | file | JoeDeveloper: only a few of them have return values, for some stuff like loadgrammar it should never fail (as long as your grammar is valid) |
| 14:59.15 | ManxPower | neoalex: As I said, I don't have any more suggestions |
| 14:59.20 | genz | PMantis: I had an issue like that too. I moved digit timeout to 2, response timeout to 60. But the big solution came when i read the loopcount parts. Can you post your IVR to pastebin? |
| 14:59.21 | PMantis | genz, Seen any of this occur? |
| 14:59.25 | fo0b | ManxPower: sure, but does svn-trunk not make a release version at some point? |
| 14:59.39 | JoeDeveloper | ManxPower, is there a command that would check my grammar before I load it? |
| 14:59.45 | ManxPower | fo0b: at some point and when it does, that patch will be part of it |
| 15:00.04 | *** join/#asterisk irule (n=irule@189.164.43.19) |
| 15:00.07 | ManxPower | JoeDeveloper: I'm still waiting for the output of "show application" |
| 15:00.07 | irule | hi there |
| 15:00.13 | fo0b | ManxPower, will it be in the 1.2 or 1.4 line? |
| 15:00.24 | JoeDeveloper | ManxPower the "show application X" is not saying anything about the return values. The doxygen indicates all these Speech commands have return values. |
| 15:00.26 | irule | does someone have the basic PPBX configured already? |
| 15:00.26 | KingCoco | hi all |
| 15:00.36 | ManxPower | fo0b: neither. It will be 1.6 |
| 15:00.56 | PMantis | genz, http://www.pastebin.ca/424016 |
| 15:01.10 | ManxPower | fo0b: OR was in the 1.4 release |
| 15:01.21 | ManxPower | I don't recall when the two branched |
| 15:01.22 | JoeDeveloper | http://pastebin.ca/424019 |
| 15:01.25 | PMantis | genz, This started happening after an upgrade to 1.4.2... 1.4.0 worked well with this. |
| 15:01.45 | irule | something aloing the features mentioned on this page? http://panasonic.co.nz/product-detail$product$679$.html |
| 15:01.45 | fo0b | ManxPower, will that patch work on a 1.2.1 release? |
| 15:01.54 | file | JoeDeveloper: loading the grammar via lumenvox.conf will tell you if anything is wrong with it (and that's the preferred way to load grammars... at startup) or loading it in the tuner |
| 15:02.02 | ManxPower | JoeDeveloper: that application does not set any variables so there is no easy way to check for success or failure. |
| 15:02.21 | ManxPower | fo0b: I doubt it. |
| 15:02.29 | KingCoco | does someone interconnect Asterisk with HiPath4000 using Sip |
| 15:02.41 | fo0b | ManxPower, How can I find out if it's in 1.4? |
| 15:02.59 | fo0b | will it be mentioned in the release notes? |
| 15:03.04 | ManxPower | fo0b: But if it is a security issue then it would have been backported to 1.2.x |
| 15:03.30 | Qwell[] | file: grammar has preferences? |
| 15:03.37 | *** join/#asterisk ltd (n=z@202-161-20-161.dyn.iinet.net.au) |
| 15:03.41 | ManxPower | file: can you look at http://bugs.digium.com/view.php?id=6334 and tell us if this patch made it into 1.2.x? |
| 15:04.21 | genz | PMantis: Is this based off of TrixBox? |
| 15:04.21 | ManxPower | Qwell: Of course! The SoCal grammer prefers the word "dude", while the Southern grammer prefers the word "y'all" |
| 15:04.31 | Qwell[] | ManxPower: gotcha |
| 15:04.34 | Qwell[] | and no, that didn't go into 1.2 |
| 15:05.18 | PMantis | genz, FreePBX created it yes. But I can do without it... installed so a client can manage their own stuff. |
| 15:05.25 | ManxPower | Qwell: Why in the world not? It seems like a security fix to me. |
| 15:05.41 | file | Qwell[]: yes, yes it does |
| 15:06.23 | PMantis | genz, I've used WaitExten before, but I don't see it used in either installation that I have, which used FreePBX. |
| 15:06.26 | MrWup | is there any function which allows me to set a variable for a particular sip phone |
| 15:06.28 | MrWup | for example... |
| 15:06.29 | PMantis | genz, Does 1.4.2 require this? |
| 15:06.30 | neoalex | is there any way to manually set status ringing for an extension |
| 15:06.30 | genz | PMantis: Have you tried increasing your response timeout? I think they changed how that is handled to include message playback |
| 15:07.14 | PMantis | genz, ok, doubling it. |
| 15:07.28 | MrWup | i got a phone SIP/101 i want to set a certain variable for it like if GetVar(SIP/101) = 'ATLUNCH' then... or SetVar(SIP/101,'INMEETING') |
| 15:07.31 | ManxPower | MrWup: setvar=varname=varvalue in the sip.conf [section] for that device. Any channels created by that device will have that channel variable set. |
| 15:07.59 | [TK]D-Fender | irule: Yes, most of us. |
| 15:08.27 | ManxPower | All of my PBXs are grand, not basic, you insensitive clod! |
| 15:08.56 | [TK]D-Fender | MrWup: No, there is no variable that will hold that per device. You need to use AstDB. |
| 15:09.05 | [TK]D-Fender | MrWup: "show function DB" |
| 15:09.05 | ManxPower | MrWup: looks like you need to set and change the variables using the Asterisk built in database, |
| 15:09.12 | MrWup | oh |
| 15:09.29 | ManxPower | MrWup: what you want to do is not what your question was. |
| 15:09.30 | *** join/#asterisk smace (n=smace_br@201.18.17.42) |
| 15:09.35 | *** part/#asterisk smace (n=smace_br@201.18.17.42) |
| 15:09.36 | *** part/#asterisk MedozasSVR (n=MedozasS@p54911513.dip0.t-ipconnect.de) |
| 15:09.39 | PMantis | genz, Nope... went up to 40 seconds, and it still fell through immeditately. |
| 15:09.59 | MrWup | [TK]D-Fender, ok so i'll use the db, thats no problem. but it comes to a crunch- namely this: can I get a sip phone to subscribe to the status of that variable |
| 15:10.35 | *** join/#asterisk wunderkin (n=kev@dslstat-ppp-95.fastq.com) |
| 15:10.41 | ManxPower | MrWup: no, you would have to handle it in the dialplan, checking the value of the variable and then deciding what to do with it. |
| 15:10.43 | MrWup | [TK]D-Fender, so basically i define a BLT programmable button on the aastra 9133i to subscribe to a hint for say extension 1234. then the hint is all about "in a meeting" or a custom action. |
| 15:11.05 | genz | PMartis: That's so strange. Mine was the same way until I changed it to this - http://pastebin.ca/424026 |
| 15:11.25 | MrWup | so that the hint telling aastra phone what to do (i.e. have the light on and steady) is based on whether the variable is set or not |
| 15:11.28 | [TK]D-Fender | MrWup: I think with the devstate patch that was created in Trunk 1.4 yes |
| 15:11.57 | [TK]D-Fender | MrWup: http://www.voip-info.org/wiki/view/Getting+app_devstate+to+work+on+Asterisk+1.4+without+rest+of+bristuff+patches |
| 15:11.58 | MrWup | [TK]D-Fender, basically i want to be able to have 5 buttons: in a meeting, out to lunch, busy with client, busy with work etc, could be anything |
| 15:12.06 | ManxPower | genz: you must have autofallthru=no in [general] |
| 15:12.14 | [TK]D-Fender | MrWup: There is an announcemt in the Asterisk.org blogs. GO READ |
| 15:12.21 | MrWup | and the user can press the button and it shows ON, and all the time it is on, people cant call em |
| 15:12.54 | MrWup | ok |
| 15:12.55 | [TK]D-Fender | Yay.... deprecated FreePBX scripts! |
| 15:13.29 | Dovid | anyone here ever install a2illing ? |
| 15:13.34 | genz | ManxPower: I'm happy with mine and I'm using the default autofallthru of yes. |
| 15:14.19 | PMantis | genz, I added a "WaitExten" line under "Background", now the IVR works as expected... |
| 15:14.20 | neoalex | ManxPower: thanks for the help anyway, I found a workarround exten => 1234,1,ringing and you were right, the problem is the provider is answering the call before processing |
| 15:14.22 | ManxPower | genz: what version of asterisk. |
| 15:14.28 | ManxPower | PMantis: what verison of asterisk are you using |
| 15:14.35 | PMantis | ManxPower, 1.4.2 |
| 15:14.36 | *** join/#asterisk vader-- (n=me@204.183.88.101) |
| 15:14.59 | PMantis | ManxPower, was working on 1.4.0 |
| 15:15.24 | ManxPower | PMantis: SOMEONE didn't do regressions testing. /me looks at the developers |
| 15:15.35 | vader-- | so whats up with Asterisk 1.4.x? whats different |
| 15:15.35 | Qwell[] | user error |
| 15:15.43 | fo0b | ManxPower, Is that functionality in 1.4? If not, and I require it am I best running off of trunk? Is trunk pretty stable? |
| 15:15.51 | ManxPower | PMantis: now you understand why I don't use low numbered point releases |
| 15:15.58 | genz | ManxPower: SVN version 59042 |
| 15:15.59 | ManxPower | fo0b: ask a developer |
| 15:16.04 | PMantis | ManxPower, Damn! a production release??? WTF? |
| 15:16.52 | Qwell[] | If you're just gonna let it timeout, what do you expect to happen? |
| 15:17.02 | *** join/#asterisk codazoda (n=chatzill@mail.hurdmanivr.com) |
| 15:17.13 | PMantis | ManxPower, the latest 1.2 (1.2.17) release works fine though.... |
| 15:17.46 | mocker | .17, 17 chances to get it right! |
| 15:17.52 | mocker | :) |
| 15:17.58 | *** join/#asterisk _VoicePulse (n=contact@unaffiliated/voicepulse) |
| 15:18.17 | badcfe | i have dtmfmode=rfc2833 in the sip peer but cant see dtmf arrive. the sender uses rfc2833 but i want to double check that without *. is there a way to do that? |
| 15:18.44 | codazoda | I'm having trouble with my TDM404B. Won't answer calls. When I install it, I get "FATAL: Error running install command for wctdm". My zaptel.conf, zapata.conf, and output from my modprobe attempts are at http://www.pastebin.ca/424039. |
| 15:18.47 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 15:19.23 | ManxPower | codazoda: if it can't install wctdm then I doubt your TDM4XXB will do ANYTHING |
| 15:20.04 | codazoda | ManxPower: Right. But, I thought the error *might* be harmless, since sometimes errors during modprobe are normal. Although I can't remember what or why. |
| 15:20.24 | codazoda | Thus, I tested. |
| 15:20.25 | giasai68 | how i can to set extensions.conf for split all incoming faxcall to email? |
| 15:20.33 | ManxPower | PMantis: 1.0.x and 1.2.x had many issues in the early releases |
| 15:20.39 | tzafrir | that error means that your /etc/zaptel.conf is incorrect . Nothing more. Basically ignore it. Or better yet: remove the line referring to wctdm in /etc/modprobe.conf |
| 15:20.42 | ManxPower | PMantis: Asteirsk is a complex piece of software. |
| 15:20.46 | codazoda | giasai68: use a pattern match. |
| 15:21.02 | ManxPower | tzanger: uh, he has a wctdm card |
| 15:21.29 | tzanger | ManxPower: eh? |
| 15:21.39 | tzanger | oh it's a tzafrir mixup :-) |
| 15:21.44 | codazoda | tzafrir: You talking to me? The error means zaptel.conf is wrong, how? |
| 15:21.50 | ManxPower | (10:18:37) codazoda: I'm having trouble with my TDM404B. Won't answer calls. When I install it, I get "FATAL: Error running install command for wctdm". My zap.... |
| 15:21.53 | tzafrir | codefreeze, yes |
| 15:22.11 | tzafrir | codazoda, what do you have on /etc/zaptel.conf ? |
| 15:22.13 | ManxPower | tzanger: two people should not have nicks starting with the same two letters 8-) |
| 15:22.14 | tzafrir | ~pb |
| 15:22.16 | jbot | somebody said pb was a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 15:22.20 | codazoda | http://www.pastebin.ca/424039 |
| 15:22.23 | *** join/#asterisk DeeJayTwo (n=deejay2@office.abi.ca) |
| 15:22.36 | tzafrir | ManxPower, three, and last one as well |
| 15:22.53 | ManxPower | evil nicks! Evil I tell ya! |
| 15:22.53 | DeeJayTwo | Beside packetization time, is there something I can do on my network to decrease latency? |
| 15:22.57 | codazoda | Yeah, well, everybody calls me codafreeze because I had a conversation with a guy with freeze in his name yesterday. ;-) |
| 15:23.00 | giasai68 | <ManxPower>: mow i done _39X. dial if is a voice call after i write fax,1,SetVar(FAXFILE=/var/spool/asterisk .... but the call will be transfer as a voice call |
| 15:24.08 | codazoda | Oh, I'm such an idiot. |
| 15:24.48 | codazoda | tzafrir: My zaptel.conf look wrong? |
| 15:26.08 | codazoda | The LED's are lit up on my TDM404B, in case that means anything. |
| 15:27.04 | tzafrir | hmm.... no |
| 15:27.24 | irule | [TK]D-Fender wouldnt it be nice to create one cool PBX configuration that we could all be working on? sort of like the way * is being built, but just a standardized dialplan we can all have in GPL as well? I am still learing but would like to find or create a real dialplan wel can all share and use |
| 15:27.25 | tzafrir | codefreeze, what distro do you use? |
| 15:28.00 | tzafrir | irule, what specific things are you missing? |
| 15:28.17 | tzafrir | codazoda, ls -l /dev/zap/ctl |
| 15:28.36 | *** join/#asterisk mocker (n=mocker@198.247.173.227) |
| 15:29.10 | codazoda | tzafrir: ls: /dev/zap/ctl: No such file or directory |
| 15:30.05 | tzafrir | ls /sys/class/zaptel/zapctl |
| 15:30.20 | codazoda | dev |
| 15:30.37 | tzafrir | right. So zaptel is loaded, but no device files were generated |
| 15:30.41 | irule | tzafrir I dont even know, I barely understand the terms used in http://panasonic.co.nz/product-detail$product$679$.html and of course a lot pf things are named differently by * pros and hobbyists, so I just think it would be cool to get together amongs ourselves and define features and start from there |
| 15:30.52 | tzafrir | This is typically done automatically by the udev rules |
| 15:30.58 | PMantis | ManxPower, Oh yeah! Very complex! That's why I needs regression testing. :) |
| 15:31.11 | PMantis | ManxPower, s/I/it/ |
| 15:31.44 | DeeJayTwo | I hear that reducing the NIC MTU can reduce RTP latency. Is it true? How's it happening? |
| 15:31.45 | tzafrir | irule, the problem is doing so without assuming too much about the PBX |
| 15:31.54 | codazoda | tzafrir: Hmmm. So maybe I compiled wrong? I'm running CentOS 4.4 w/ udev. I did 'make install-udev' during the install. Maybe in the wrong order or something? |
| 15:31.56 | DeeJayTwo | s/hear/heard/ |
| 15:32.19 | tzafrir | grep zap /etc/udev/rules.d/* |
| 15:32.39 | codazoda | There are rules listed. |
| 15:32.54 | tzafrir | from what file(s)? |
| 15:33.29 | MrWup | i assume after putting app_devstate in asterisk.... i need to rebuild asterisk? |
| 15:33.29 | codazoda | http://www.pastebin.ca/424061 |
| 15:33.35 | MrWup | in /apps |
| 15:34.41 | irule | can someone point me to a regular PBX diaplan in case it is online? :s |
| 15:34.58 | codazoda | I've done this several times, and it's all worked fairly easily so far. This is about my 4th similar install. One major difference, I didn't install CentOS on this machine, a co-worker did. And, it's 64-bit. In case that's a problem. |
| 15:36.02 | MrWup | is it possible to use playback with files other than GSM? |
| 15:36.07 | Qwell[] | MrWup: sure |
| 15:36.13 | MrWup | cause for example, all my SIP phones use alaw |
| 15:36.15 | *** join/#asterisk tuan_modulis (n=chatzill@3-82-252-216-static.enter-net.com) |
| 15:36.32 | MrWup | and im doing playback wtih GSM files and while my sip to sip voice quality sounds good, playback sounds shit |
| 15:37.06 | MrWup | Qwell, how do i do it? just put a .wav in the sounds dir and playback it? or put a .alaw file in and play it? im a noob to this i dont understand much |
| 15:37.37 | tuan_modulis | u can play wav files if you want |
| 15:37.56 | *** join/#asterisk s34n (n=chatzill@ip-206-159-190-125.mvdsl.com) |
| 15:38.13 | tuan_modulis | playback doesnt use the suffix.... but make sure the wav is in asterisk format |
| 15:38.31 | MrWup | tuan_modulis, how do i know what asterisk format is? |
| 15:38.34 | codazoda | tzafrir: No more idea's, eh? |
| 15:38.56 | s34n | on a very low muscle server, if I can control codecs end-to-end, which single codec would tax my cpu the least, or does it matter if I never have to transcode? |
| 15:39.32 | tuan_modulis | MrWup: google sox and asterisk... im sure u can find the command somewhere for that |
| 15:39.35 | ManxPower | MrWup: "show translations" |
| 15:39.48 | tzafrir | codazoda, two things to try: (1) reload/restart udevd (probably something like /etc/init.d/udev reload) and... |
| 15:39.51 | ManxPower | if it has a number in the collumn then it can play that format |
| 15:40.06 | MrWup | oh i thought sox was only used for converting wav to gsm |
| 15:40.21 | MrWup | i mean i can take a standard wav with sox and convert it into a wav that asterisk likes? |
| 15:40.26 | ManxPower | MrWup: sox can be used to convert many formats to many other formats. |
| 15:40.30 | MrWup | oh |
| 15:40.35 | tzafrir | codazoda, (2)rmmod wctdm; modprobe wctm and see if suddenly device files were generated |
| 15:40.42 | ManxPower | MrWup: "man sox" |
| 15:40.53 | MrWup | ManxPower, is there a list of formats which asterisk supports? |
| 15:40.54 | PMantis | genz, Thans for the help... for now, I hacked fpbx to include the "WaitExten" line in all IVRs |
| 15:40.58 | MrWup | so i know what to convert to? |
| 15:41.11 | *** join/#asterisk Deeewayne (i=dwayne@nat/digium/x-cc97a6de0c58e743) |
| 15:41.12 | ManxPower | MrWup: YES! SHOW TRANSLATIONS SHOWS THE FORMATS ASTERISK SUPPORTS |
| 15:41.29 | ManxPower | (10:39:26) ManxPower: MrWup: "show translations" |
| 15:41.55 | MrWup | oh |
| 15:41.58 | MrWup | in the console? |
| 15:42.04 | ManxPower | MrWup: Asterisk expects 8Khz mono sound files. |
| 15:42.05 | ManxPower | I don't know the number of bits, maybe 16 bits |
| 15:42.13 | ManxPower | MrWup: in the Asterisk CLI |
| 15:42.18 | MrWup | ok thanks |
| 15:42.29 | ManxPower | ulaw is basically wav |
| 15:43.33 | MrWup | oh |
| 15:43.43 | MrWup | in europe we apparently use alaw |
| 15:43.51 | MrWup | ulaw and alaw are the same quality right? |
| 15:43.57 | ManxPower | MrWup: correct. |
| 15:44.14 | MrWup | so whats the difference meaning that in the US they use ulaw and europe alaw |
| 15:44.22 | MrWup | just different names or is there more to it? |
| 15:44.33 | *** join/#asterisk djs_2_6 (n=DJS@cpe-071-077-048-198.nc.res.rr.com) |
| 15:44.43 | coppice | they are different |
| 15:44.52 | ManxPower | MrWup: I have never needed to know the differences. |
| 15:45.19 | coppice | ulaw and alaw are similar lossy compression schemes, but they are not identical |
| 15:45.30 | giasai68 | hoa i can detect with asterisk if a call is a voice call or fax call? because if is a voice call i need to traslate to an ip if is a fax i need to have it in a file via email |
| 15:46.07 | ManxPower | coppice: Oh telco guru, do you have any idea what might cause echo on analog lines with Asterisk. i.e. all TDM bridged. |
| 15:46.48 | coppice | isn't that a normal thing to have, if you don't have good echo cancellers? |
| 15:46.53 | s34n | what is the least cpu intensive codec? |
| 15:47.15 | coppice | ulaw and alaw use almost nothing |
| 15:48.01 | *** join/#asterisk zotz (n=zotz@24.244.163.157) |
| 15:48.08 | ManxPower | coppice: not with all TDM |
| 15:48.26 | codazoda | tzafrir: If I "rmmod wctdm; modprobe wctdm", I don't get an error. But, if I "rmmod wctdm; rmmod zaptel; modprobe zaptel; sleep 3; modprobe wctdm" I still get it. |
| 15:48.32 | coppice | why? |
| 15:48.41 | codazoda | I can't figure out how to reload udev. Maybe a reboot is in order... |
| 15:48.57 | tzafrir | codazoda, as I said, there is really no need to run ztcfg as a post-install action of the modprobe |
| 15:49.06 | s34n | coppice: so, G.711u or G711a, you mean? |
| 15:49.06 | giasai68 | how i can detect with asterisk if a call is a voice call or fax call? because if is a voice call i need to traslate to an ip if is a fax i need to have it in a file via email |
| 15:49.14 | ManxPower | coppice: because the TDM latency is low enough you would not normally hear the echo. |
| 15:49.21 | coppice | s34n: yes |
| 15:50.04 | s34n | coppice: If I never have to transcode, will it matter? |
| 15:50.07 | codazoda | tzafrir: Not sure I understand that last comment. But, now I think I'm missing something, since I never ran ztcfg. Does modprobe zaptel do so? |
| 15:50.13 | tzafrir | codazoda, just run ztcfg later |
| 15:50.37 | s34n | for instance if my trunk, my phones, my prompts, and my voicemail are all the same codec...? |
| 15:50.40 | tzafrir | grepo ztcfg /etc/modprobe.conf # to see where it is run |
| 15:51.04 | coppice | s34n: if you don't transcode, the CPU will obviously spend zero time transcoding whichever codec you use |
| 15:52.07 | s34n | coppice: so if I set all those to the same codec, can I completely eliminate that load from the * server? |
| 15:52.09 | fo0b | why should I choose 1.2.x over 1.4.x? |
| 15:52.29 | s34n | coppice: or is there some codec processing * can never escape? |
| 15:52.37 | codazoda | tzafrir: Odd. A full reboot solved the problem. Not sure why. |
| 15:52.52 | s34n | fo0b: who said you should? |
| 15:52.59 | mihinomenest | I have more than one SIP account that I want to use for outbound calling, what should I be looking for on the voip-info wiki? |
| 15:53.00 | codazoda | tzafrir: Thanks for your help. |
| 15:53.15 | tzafrir | codazoda, you can replicate it with: rmmod wctdm zaptel; modprobe wctdm |
| 15:53.21 | fo0b | s34n, nobody. I was just wondering what reasons there are NOT to go with 1.4? |
| 15:53.24 | tuan_modulis | fo0b: I'm using 1.4 in hopes of when asterisknow becomes stable |
| 15:53.35 | tuan_modulis | but some people consider it bleeding edge still |
| 15:53.38 | tzafrir | codazoda, assuming you don't have any other zaptel modules loaded |
| 15:53.42 | coppice | s34n: if it cannot turn the audio to plain linear samples there is no way to do things like tone detection. if you don't need that, there will be no CPU use |
| 15:53.54 | codazoda | tzafrir: I had tried that several times. Maybe it was udev that needed restarting. |
| 15:54.06 | *** part/#asterisk PMantis (n=pmantis@cpe-74-65-53-178.rochester.res.rr.com) |
| 15:57.08 | genz | Is there a setting that controls the recording volume? Some of our voicemail recordings are very loud. |
| 15:57.37 | genz | fo0b: I'm using 1.4 and am happy with it. |
| 15:57.56 | MrWup | hmf |
| 15:58.05 | MrWup | ok ive compiled asterisk with the devstate app |
| 15:58.11 | MrWup | core show applications lists devstate |
| 15:58.24 | MrWup | but when i try and use devstate in the extensions.conf it says function devstate is not registered |
| 16:03.12 | *** join/#asterisk harleya (n=harleya@207.108.166.2) |
| 16:10.50 | tuan_modulis | http://www.voip-info.org/wiki/view/Getting+app_devstate+to+work+on+Asterisk+1.4+without+rest+of+bristuff+patches |
| 16:10.57 | tuan_modulis | that's what u did? |
| 16:14.01 | *** join/#asterisk vgster (n=vgster@host217-45-221-53.in-addr.btopenworld.com) |
| 16:16.01 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 16:17.59 | *** join/#asterisk fakhir (i=fakhir@unaffiliated/fakhir) |
| 16:18.13 | MrWup | beirut! |
| 16:20.08 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 16:22.22 | giasai68 | how i can detect with asterisk if a call is a voice call or fax call? because if is a voice call i need to traslate to an ip if is a fax i need to have it in a file via email |
| 16:23.07 | ManxPower | giasai68: the way shown in the zapata.conf.sample config file, or the way shown on the Wiki for NVFaxDetect |
| 16:23.57 | MrWup | [TK]D-Fender, u around? |
| 16:24.00 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 16:24.26 | *** part/#asterisk ming_zym (n=ming_zym@124.254.56.42) |
| 16:25.33 | [TK]D-Fender | MrWup: yes, what is it? |
| 16:26.04 | giasai68 | yes this is if i want detect fax incoming on zap channels not if i get call in sip rigth? |
| 16:26.28 | MrWup | [TK]D-Fender, i dont know if you have any experience with this, but about that Devstate thing... I went to the page about how to get it to work... put the app_devstate.c into my apps... edited the make menuconfig thing, then edited the modules tree, then make clean then make then make install |
| 16:26.49 | MrWup | [TK]D-Fender, however when i call Devstate from extensions.conf it says function Devstate is not registered |
| 16:26.50 | [TK]D-Fender | MrWup: Check the * blogpost I told you about... better instructions and SAMPLES |
| 16:27.11 | [TK]D-Fender | MrWup: And tehre is a FUNCTION, and perhaps an APPLICATION as well |
| 16:27.28 | MrWup | oh |
| 16:27.51 | mmartinn | Is anyone here using asterisk_run_loop in Gentoo's Asterisk init script? I can't get it to start Asterisk correctly |
| 16:27.58 | MrWup | [TK]D-Fender, thats where i went to |
| 16:28.12 | [TK]D-Fender | MrWup: paste the link. |
| 16:28.25 | MrWup | http://www.voip-info.org/wiki/view/Getting+app_devstate+to+work+on+Asterisk+1.4+without+rest+of+bristuff+patches |
| 16:29.38 | MrWup | and |
| 16:29.39 | MrWup | http://www.asterisk.org/node/48325 |
| 16:29.52 | MrWup | and i tried those examples. still says function Devstate not registered |
| 16:29.54 | *** join/#asterisk Mercestes (n=Merceste@rrcs-71-41-157-70.sw.biz.rr.com) |
| 16:31.19 | MrWup | *CLI> [Apr 4 17:33:51] ERROR[16833]: pbx.c:1523 ast_func_read: Function Devstate not registered |
| 16:31.29 | [TK]D-Fender | MrWup: well I can tell you that you should not have an APP for app_devstate.so |
| 16:31.52 | [TK]D-Fender | MrWup: In trunk you should have the option for func_devstate which needs to be compiled in. |
| 16:32.34 | MrWup | trunk? |
| 16:32.45 | [TK]D-Fender | MrWup: SVN trunk. |
| 16:32.55 | MrWup | i dont think i have that |
| 16:33.14 | [TK]D-Fender | MrWup: Then I don't think you're getting very far :) |
| 16:33.22 | MrWup | heh |
| 16:33.35 | [TK]D-Fender | MrWup: Now I have not attempted this PERSONALLY yet as I have hardly touched 1.4 at all. YMMV |
| 16:33.46 | MrWup | trunk is in addons? |
| 16:34.42 | mocker | So is anyone recommending 1.4 for production? |
| 16:34.45 | mocker | I've heard mixed things. |
| 16:36.11 | *** join/#asterisk samy_b1 (n=baind@2001:49f0:1000:0:0:0:0:7) |
| 16:36.15 | samy_b1 | hi al l |
| 16:36.47 | samy_b1 | did some one know why spandsp web is down |
| 16:37.08 | samy_b1 | and is there any other way for me to resive and send fax ? |
| 16:40.27 | MrWup | what on earth is svn trunk |
| 16:41.11 | samy_b1 | I don't know |
| 16:42.26 | tzafrir | coppice, new content for http://soft-switch.org ? |
| 16:42.43 | coppice | nope. failed servers |
| 16:43.03 | tzafrir | where was that mirror? |
| 16:43.08 | red9012 | anyone in here has an updated queue realtime table definition? |
| 16:44.10 | coppice | I thought I was getting sneaky having www.soft-switchorg and soft-switch.org on different machines. today they have both failed :-\ |
| 16:46.11 | samy_b1 | well |
| 16:46.34 | menfin | something very strange happening with my asterisk ipbx |
| 16:46.38 | samy_b1 | i'm offering free mirror space if they like |
| 16:46.52 | *** join/#asterisk Beirdo (n=gjhurlbu@unaffiliated/beirdo) |
| 16:47.07 | menfin | it's running perfectly ! |
| 16:47.16 | menfin | it's marvelous |
| 16:47.24 | menfin | You know what ? |
| 16:47.30 | menfin | I'm happy ! |
| 16:47.35 | menfin | :) |
| 16:47.43 | samy_b1 | i'm happy that you are happy |
| 16:50.37 | samy_b1 | did some one try to use any new vercions on spandsp ? |
| 16:50.54 | samy_b1 | is asterisk 1.4 comes with it? |
| 16:52.47 | menfin | what is spandsp please ? |
| 16:53.35 | Mercestes | menfin: It's this.....thing that rxfax and txfax uses for faxing. dunno what it is exactly |
| 16:53.38 | MrWup | hmf |
| 16:53.41 | Mercestes | It's a module, technically. |
| 16:53.41 | samy_b1 | that is a plugin that allow you to sned and recive fax |
| 16:53.47 | MrWup | well i got func_devstate.c and now it wont compile |
| 16:56.16 | menfin | Mercestes, i see, is it under gpl ? |
| 16:56.49 | samy_b1 | yes it is |
| 16:57.00 | samy_b1 | but them URL is down |
| 16:57.05 | Mercestes | samy_b1 sounds like he knows more about it than I do. |
| 16:57.06 | menfin | interesting |
| 16:57.50 | samy_b1 | well i used it for some time with freePBX |
| 16:58.20 | samy_b1 | and i like to ugrade to 1.4.x |
| 16:58.35 | menfin | is this module able to send fax over voip ? |
| 16:58.38 | samy_b1 | but i'm not sure is it going to work at al lor not |
| 16:58.43 | samy_b1 | yes |
| 16:58.48 | samy_b1 | menfin |
| 16:59.12 | menfin | (excuse my poor english i'm a little frenchie) |
| 16:59.33 | menfin | (Poor lonesome frenchie far away from home ..) |
| 17:00.01 | *** join/#asterisk nxspnt (n=noaxispo@outbound.tritonmanagementgroup.com) |
| 17:00.09 | samy_b1 | my english is not betther then yours i'm bulgarion |
| 17:00.17 | menfin | i'm looking for some documentation about this module |
| 17:01.01 | *** join/#asterisk myiagy (n=myiagy@200.175.61.250.static.gvt.net.br) |
| 17:01.19 | samy_b1 | http://web.archive.org/web/20060519111827/http://www.soft-switch.org/ |
| 17:01.26 | *** part/#asterisk nxspnt (n=noaxispo@outbound.tritonmanagementgroup.com) |
| 17:01.26 | samy_b1 | menfin try that |
| 17:01.59 | MrWup | [TK]D-Fender, i got func_devstate.c from the svc trunk and put it in /funcs and tried to compile, i just get errors like "95 error: return type is an incomplete type |
| 17:01.59 | MrWup | " |
| 17:02.32 | *** join/#asterisk jterrero (n=elguy@200.42.250.233) |
| 17:02.45 | [TK]D-Fender | MrWup: Check the mailing lists and ask russellb if he rouses |
| 17:03.09 | jterrero | anyone want to make a quick 60 bucks helping me with a 10 minute config, msg me |
| 17:03.23 | Qwell[] | jterrero: ask in the channel, I'm sure somebody will help |
| 17:04.05 | jterrero | need someone to do a config from scratch, extensions.conf and sip.conf, |
| 17:04.22 | *** join/#asterisk falz (n=falz@proxy.supranet.net) |
| 17:04.41 | falz | good day. from what I see, the tdm800x cards are only supported with zap 1.4? |
| 17:04.55 | menfin | thanks samy_b1 |
| 17:05.08 | menfin | what time is it for you ? |
| 17:05.27 | Qwell[] | falz: latest 1.2 also |
| 17:05.43 | *** join/#asterisk rene- (n=rene@200.34.66.137) |
| 17:05.46 | falz | Qwell: oh? hmm. perhaps it's just a freebsd issue I'm having, or that port is behind |
| 17:05.52 | Qwell[] | likely |
| 17:06.12 | Qwell[] | it probably won't work on freebsd for some time, actually |
| 17:06.14 | falz | I was trying hard to get to freebsd on this |
| 17:06.21 | falz | ? |
| 17:06.33 | Qwell[] | 1.4 or otherwise |
| 17:06.33 | falz | I see he's porting 1.4, but are there other fundimental changes in it? |
| 17:06.40 | Qwell[] | yes |
| 17:06.50 | Qwell[] | the drivers are all rewritten |
| 17:06.54 | falz | oh snap. |
| 17:06.56 | falz | that would do it |
| 17:06.57 | Qwell[] | and not all hardware is supported |
| 17:07.00 | *** join/#asterisk djbrh (n=djb@216-166-211-66.clec.commercial.madisonriver.net) |
| 17:07.13 | rene- | hello, how much power does the CO provides in a pots line? |
| 17:07.43 | s34n | coppice: is IAX CPU intensive? |
| 17:07.44 | coppice | the power to talk to people |
| 17:08.07 | rene- | coppice: heh |
| 17:08.07 | rene- | how rhetorical |
| 17:08.11 | falz | Qwell: thanks for the heads up. debian it will have to be then. damn having dosens of freebsd servers and one linux |
| 17:08.25 | *** join/#asterisk codefreeze (n=steve_mu@216.166.159.235) |
| 17:08.57 | *** join/#asterisk coppice (n=chatzill@90.203.17.210.dyn.pacific.net.hk) |
| 17:09.35 | rene- | ok, the ar1688 chip claims to use less than 90mw in operation, maybe a low spec P phone+modem combo could run on such little power? |
| 17:10.19 | *** join/#asterisk notoriousrab1982 (n=notoriou@207.47.34.74.static.nextweb.net) |
| 17:12.45 | [TK]D-Fender | s34n: No more than SIP+RTP |
| 17:15.06 | djbrh | so I've got a fresh trixbox install that's also been brought fully up to date. configured two SIP phones to have extensions and enabled voicemail for them. the extensions can call one another, but any entry into the vm system just results in an open call with no output from the vm system. either calling and letting vm answer or dialing into vm with *97 both just sit there. |
| 17:15.15 | s34n | [TK]D-Fender: I'm trying to cram * + dhcp + tftp into 800MHz, and I want to be reasonably confident it will work before I buy the hardware. |
| 17:15.17 | djbrh | I've done this before with no problem. |
| 17:15.22 | djbrh | but this install is borked |
| 17:15.27 | djbrh | anyone know where to start looking? |
| 17:16.17 | Qwell[] | djbrh: See channel topic |
| 17:16.31 | djbrh | ah |
| 17:16.31 | djbrh | thanks |
| 17:16.38 | [TK]D-Fender | s34n: you could pump HUNDREDS of calls at a time through that system..... what will kill you is TRANSCODING. What codecs are you planning on using? dhcp + TFTP = less than nothing |
| 17:16.40 | *** part/#asterisk djbrh (n=djb@216-166-211-66.clec.commercial.madisonriver.net) |
| 17:17.13 | s34n | [TK]D-Fender: I will keep the codec the same throughout |
| 17:17.46 | s34n | [TK]D-Fender: As long as trunk, prompts, phones, and voicemail are the same, it shouldn't matter the codec, right? |
| 17:17.51 | [TK]D-Fender | s34n: Then there should be no problem. |
| 17:17.57 | [TK]D-Fender | s34n: What codec? |
| 17:18.11 | s34n | [TK]D-Fender: probably 711u |
| 17:18.23 | [TK]D-Fender | s34n: No issue at all. |
| 17:19.09 | *** join/#asterisk jm|laptop (n=jm|home@dilbert.jamiem.com) |
| 17:19.43 | Voice2 | im looking for a web based sip phone any idea ? |
| 17:19.55 | Voice2 | https://www.gizmocall.com/ |
| 17:19.57 | Voice2 | somethign like this |
| 17:19.58 | s34n | [TK]D-Fender: even 729 should be safe, but I think 711 will be more voice email friendly |
| 17:20.28 | *** join/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 17:21.09 | [TK]D-Fender | s34n: safest bet, and transcode from G.711 to GSM (typical recording codec) is very nominal load. |
| 17:21.28 | brodiem | can anyone recommend a good "broadband-grade" router that will offer QoS for SIP/media? |
| 17:22.49 | s34n | [TK]D-Fender: it occurred to me though that IAX might pose a fly in the ontment. |
| 17:24.00 | brodiem | and I am not interested in WRT's w/ modded firmware |
| 17:24.01 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 17:26.09 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 17:27.12 | philippel | question: are there any known issues with running multiple asterisk instances on the same box, once you've adjusted all the various bindaddr, changed manager ports on each - etc. ? |
| 17:27.17 | [TK]D-Fender | s34n: And why is that? |
| 17:27.26 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 17:27.57 | s34n | [TK]D-Fender: I just don't know what resources IAX requires. |
| 17:28.13 | [TK]D-Fender | s34n: you clearly have little imagination :) |
| 17:29.12 | [TK]D-Fender | s34n: All IAX2 is, is a single-port encapsulation of the RTP data + equivalent of SIP signalling (non-issue). It is smaller in data saize than SIP, and can save on bandwidth |
| 17:29.12 | s34n | [TK]D-Fender: I imagined that it could be as much effort to "transcode" to IAX as to transcode between codecs |
| 17:29.34 | [TK]D-Fender | s34n: every bit as much. not significantly more OR less. |
| 17:31.38 | s34n | [TK]D-Fender: so if everything else is 711 and my trunk is IAX, 800MHz should still be fine, right? |
| 17:31.55 | [TK]D-Fender | s34n: absolutely |
| 17:32.19 | s34n | [TK]D-Fender: thx |
| 17:33.32 | *** join/#asterisk chefrs (n=joe@c-24-8-226-145.hsd1.co.comcast.net) |
| 17:33.39 | chefrs | I have a program running that I can telnet into when I'm SSH'd into the same box but I can't access that program from outside the box. It's for my AsterFax setup. |
| 17:33.45 | chefrs | Anyone know how to fix? |
| 17:34.04 | brodiem | chefrs the "program" is only listening on 127.0.0.1 |
| 17:34.12 | chefrs | It needs to listen on the servers IP, correct? |
| 17:34.21 | chefrs | So instead of the lo, it should be the eth0 IP. |
| 17:34.26 | brodiem | chefrs or you have iptables/fw rules disallowing it from the outside |
| 17:34.28 | brodiem | yes |
| 17:34.44 | brodiem | chefrs i assume the "program" is probably sendmail? |
| 17:35.02 | MrWup | hmf |
| 17:35.11 | MrWup | why the hell to people publish source that wont compile |
| 17:35.14 | chefrs | No, AsterFax. I had to switch it's default port to not use sendmail |
| 17:35.16 | MrWup | damned func_devstart |
| 17:35.32 | MrWup | devstate even |
| 17:35.55 | *** join/#asterisk Goodjoke (n=chatzill@mail.theenergynetwork.com) |
| 17:38.11 | Goodjoke | i am a total newb... ready for a new question? |
| 17:38.36 | Mercestes | Goodjoke, 42. |
| 17:39.07 | Goodjoke | ? |
| 17:39.07 | *** join/#asterisk af_ (n=getsmart@81-174-9-35.f5.ngi.it) |
| 17:39.11 | chefrs | lol |
| 17:39.19 | Mercestes | Goodjoke, What's the question? |
| 17:39.57 | Goodjoke | i know with the older versions... i can have two different digital receptionist messages.. one during biz hours and one during non biz hours |
| 17:40.18 | Goodjoke | any way to have different dig rec. messages based on time of day? |
| 17:40.43 | brodiem | sounds like you're using freebpx.. |
| 17:40.46 | Goodjoke | for example, play one greeting 'good morning..' from 8am-12pm' |
| 17:40.59 | chefrs | Goodjoke: Yes. Easily. |
| 17:41.27 | Mercestes | GoodJoke: cmd GotoIfTime is what you want. |
| 17:41.28 | Goodjoke | i've only used an odler ver of *@home |
| 17:41.49 | Mercestes | yea, that's freepbx/trixbox stuff |
| 17:42.01 | Goodjoke | ok.. thanks |
| 17:42.02 | [TK]D-Fender | Goodjoke: A@H, Trixbox, FreePBX, AMP, are NOT supported here |
| 17:42.02 | Mercestes | what are you using now? |
| 17:42.14 | [TK]D-Fender | ~freepbx |
| 17:42.28 | jbot | rumour has it, freepbx is unable to be supported here. It's a complex piece of dialplan and GUI, and can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 17:42.39 | *** join/#asterisk Meaty (n=meaty3@office.abi.ca) |
| 17:43.28 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 17:43.29 | PakiPenguin | hi |
| 17:43.34 | Mercestes | hi |
| 17:45.25 | *** join/#asterisk mindCrime (n=chatzill@216.187.233.42) |
| 17:46.00 | MrWup | http://www.asteriskpbx.org/doxygen/trunk/func__devstate_8c.html |
| 17:46.15 | MrWup | in the dependencies diagram does anyone know what the difference between the red and black boxes are? |
| 17:46.25 | MrWup | how would i go about compiling this source properly? |
| 17:52.28 | *** join/#asterisk ToyMan (n=Stuart@72.168.167.241) |
| 17:56.28 | *** join/#asterisk gbodemantv (n=gbodeman@corpex.pivotal.televerde.com) |
| 17:56.31 | gbodemantv | hey all |
| 17:56.42 | gbodemantv | so anyone else have 1.2.17 crash |
| 17:56.53 | gbodemantv | I have has a crash every other day |
| 17:57.11 | gbodemantv | with multiple chan_sip.c: SIP message could not be handled, bad request: 4514BBC4-1DD2-11B2-9F2F-FAFD8E12B3F6@10.210.20.95 |
| 17:57.14 | gbodemantv | errors |
| 18:00.08 | Hmmhesays | da da da da da gnome is a pain in the ass |
| 18:01.49 | brodiem | can anyone recommend a good "broadband-grade" router that will offer QoS for SIP/media (not WRT w/ modded firmware)? |
| 18:02.02 | Hmmhesays | whats wrong with the wrt's? |
| 18:02.21 | brodiem | Hmmhesays nothing but it needs to be something drop-shippable |
| 18:02.43 | Hmmhesays | ahh |
| 18:03.39 | Hmmhesays | bah, I can't figure out why gnome is freaking out |
| 18:04.14 | brodiem | ugh |
| 18:06.05 | Hmmhesays | i'm trying to get my multi-skype gateway running |
| 18:06.05 | brodiem | Hmmhesays and it requires gnome? |
| 18:06.05 | Hmmhesays | requires some wm |
| 18:06.05 | Hmmhesays | gnome is just what is on here |
| 18:06.05 | brodiem | ah |
| 18:06.20 | Hmmhesays | basically it runs multiple instances of skype in different sessions |
| 18:06.20 | brodiem | xinit /path/to/skypetool |
| 18:07.53 | Hmmhesays | well I have a script that starts multiple vnc servers |
| 18:08.09 | Hmmhesays | from there I have to modify PATH to include my path to skype |
| 18:08.29 | Hmmhesays | i'm doing that with .bash_profile for each user |
| 18:09.17 | notoriousrab1982 | polycom ip501: restore factory settings by pressing 4,6,8 and *, anyone found that when doing this, the original settings are still present? |
| 18:09.41 | brodiem | Hmmhesays why not just run multiple X servers/sessions |
| 18:11.37 | Hmmhesays | the vnc script is starting multiple X sessions |
| 18:11.39 | *** join/#asterisk lerat (n=dnormand@bas2-montreal19-1178061804.dsl.bell.ca) |
| 18:11.45 | brodiem | oh ok |
| 18:12.16 | lerat | does anyone know how to resolve this message : Got SIP response 481 "Call/Transaction Does Not Exist" back from XXX.XXX.XXX.XXX |
| 18:12.38 | lerat | or what it means??? |
| 18:13.23 | *** join/#asterisk CunningPike (n=CunningP@dhcp-10-234.district.north-van.bc.ca) |
| 18:14.33 | *** join/#asterisk boojit (n=boojit@gw.carter.to) |
| 18:14.52 | boojit | hi: any professional asterisk consultants in here? |
| 18:15.39 | Voice2 | yes |
| 18:16.00 | Voice2 | You need support ? |
| 18:16.11 | lerat | Voice2 can you answer my question??? |
| 18:16.17 | boojit | yes, sort of. I'm looking for a vendor for a specific, weird application |
| 18:16.25 | Voice2 | hehea sk away |
| 18:16.37 | boojit | I think asterisk would be a good fit but I don't know enough about PBX stuff to be sure |
| 18:16.54 | irule | what does it mean? [Apr 4 07:13:41] WARNING[3324]: res_musiconhold.c:947 local_ast_moh_start: No class: yes |
| 18:16.56 | boojit | Voice2: can I msg you please? |
| 18:17.01 | Voice2 | sur |
| 18:17.11 | Voice2 | sdoubt i can help on 3rd party weird apps but |
| 18:17.27 | Voice2 | didnt finish my weird apps certification |
| 18:17.43 | Voice2 | but finished the goold ole apps and the super good ones |
| 18:19.13 | lerat | Voice2 , can you answer my question ? |
| 18:21.10 | lerat | anybody know what a 481 response call/transaction... means ??? |
| 18:23.29 | *** join/#asterisk ManxPower (n=manxpowe@dpc67142183150.direcpc.com) |
| 18:23.47 | ManxPower | Qwell[]: pimng |
| 18:23.56 | Qwell[] | ? |
| 18:24.23 | *** join/#asterisk billytwowilly (n=chris@S01060016b649355d.ed.shawcable.net) |
| 18:24.33 | ManxPower | Do you know who might be able to answer a quick question about echocancelwhentdmbridged |
| 18:24.46 | Qwell[] | no idea |
| 18:25.11 | billytwowilly | doing a ping -c 30 to the did provider and getting a ping between 70 and 110 will probably be too much jitter eh? |
| 18:26.11 | Qwell[] | billytwowilly: ping is icmp - rtp is udp |
| 18:26.17 | Qwell[] | They would be QoSed very differently |
| 18:26.37 | ManxPower | I have a bunch of FXS and a few FXO ports. I don't want to do EC when the call is FXS<->FXS, but I do when the call is FXO<->FXS. I set echocancel=no for the FXS and echocancel=yes and echocancelwhenbridged=yes for the FXO. My question is will a call have EC of the call is FXO<->FXS in this configuration? |
| 18:27.21 | ManxPower | the reason for not wanting EC on FXS<->FXS is because I'm using the HPEC and am worried about CPU. |
| 18:28.21 | Hmmhesays | how do I go about running a script after gnome starts and the user is logged in |
| 18:29.08 | brodiem | Hmmhesays how are you starting X/gnome |
| 18:30.12 | billytwowilly | qwell so I should do what to test it? |
| 18:30.15 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 18:30.17 | billytwowilly | ping -rtsp ? |
| 18:30.54 | Hmmhesays | I have a script that starts vnc for all the drones I have |
| 18:31.34 | ManxPower | Qwell[]: It is really weird, but the HPEC actually removed the hum I have been having on the lines. |
| 18:31.44 | Qwell[] | nice |
| 18:31.45 | irule | can someone recommend a good motherboard for a solid asterisk? the book mentions its importance but no current models are mentioned |
| 18:32.08 | ManxPower | at least on the line I was testing. |
| 18:32.36 | *** join/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 18:33.17 | coppice | that's not weird at all |
| 18:33.51 | ManxPower | coppice: I was HOPING it would, as it was a fairly low frequency hum |
| 18:34.02 | *** join/#asterisk joe (n=nnnjsaue@ip66-107-33-195.z33-107-66.customer.algx.net) |
| 18:34.05 | ManxPower | I suspect harmonics of 60Hz (AC power) |
| 18:34.40 | coppice | a good echo canceller *must* block DC, and they usually filter such that they will kill 50, 60, 150 and 180 pretty effectively too |
| 18:35.18 | *** join/#asterisk mvanbaak (i=michiel@82.103.132.43) |
| 18:35.27 | ManxPower | coppice: Ah! one of my lines had hum around 180. The other one was around 500 or so, I should check that one too |
| 18:35.38 | mvanbaak | ha ! |
| 18:36.15 | notoriousrab1982 | anyone found that with polycom ip501s, when restored to factory settings, they still keep some data added previously, like a password for a sip extension |
| 18:36.42 | notoriousrab1982 | it seems some data once in the phone, you cannot get rid of it, anyone had this problem? |
| 18:37.05 | [[blah]asfd | i have a phone that shows up in the sip show peers as connected, and no one is on the phone, but from the cli when I type "dial 3328@default" I get the following errors: http://pastebin.ca/424323. if I power off the phone, then i get this message... So I know that the status of the phone is having some affect. could anyone please assist? |
| 18:37.07 | *** join/#asterisk [hC] (n=hardcore@S01060016b61c8983.vf.shawcable.net) |
| 18:37.24 | ManxPower | notoriousrab1982: I have never seen that. |
| 18:37.40 | ManxPower | notoriousrab1982: unless of course you have a config file for that phone on the prvisioning server |
| 18:37.58 | irule | http://www.discountvoipoutlet.com/store/product.php?productid=53 does the TDM400P require modules? or can I get it with no modules and start using it? |
| 18:38.17 | ManxPower | irule: it is prett useless with no modules |
| 18:38.26 | notoriousrab1982 | yeah, its weird, I am using an old phone that used to be provisioned from a ftp server, i am trying to clear it back to factory settings and provision it locally to get it to register, i keep getting on the console when it tries to register, the old password from the previous settings |
| 18:38.50 | ManxPower | notoriousrab1982: you removed the config file from the ftp server? |
| 18:38.57 | notoriousrab1982 | ie i program in to the phone locally password eg 123, and it still tries to register on the asterisk server using the old password |
| 18:39.00 | lerat | Any * professionnal consultant here ? |
| 18:39.14 | ManxPower | lerat: many. |
| 18:39.24 | lerat | i need just one |
| 18:39.29 | Voice2 | can llumenvox do french ? |
| 18:39.29 | notoriousrab1982 | manxpower, I restored factory settings, it gets rid of any ftp server settings for bootload, when you go into setup during boot, you can see that it is not pointed at an ftp server |
| 18:39.35 | ManxPower | most of them require a lengthy interview process before accepting a new client |
| 18:39.38 | ManxPower | notoriousrab1982: NO! |
| 18:40.02 | lerat | what kind of interview? |
| 18:40.19 | *** join/#asterisk mvanbaak (i=michiel@82.103.132.43) |
| 18:40.23 | ManxPower | notoriousrab1982: The phone can get it's boot server via a DHCP option in the DHCP response packet. At least Cisco phones will, if they don't know anything else, will just try contacting the IP of the DHCP server and try to download its configs from there. |
| 18:40.32 | Mercestes | The "make sure this joker will actualy *PAY* me" interview. |
| 18:40.52 | Mercestes | that reminds me...I have a PBX to shutdown from someone who hasn't paid me. =/ |
| 18:41.06 | Voice2 | Mercestes, remove sip config |
| 18:41.07 | Voice2 | lol |
| 18:41.16 | Mercestes | Voice2: ? |
| 18:41.20 | ManxPower | Mercestes: Mine usually tries to find out 1) will they pay, 2) are they idiots 3) did they do something so stupid I can't work with them and 4) are their requirements a good match for my skills. |
| 18:41.25 | Voice2 | ".I have a PBX to shutdown from someone who hasn't paid me" |
| 18:41.33 | notoriousrab1982 | manxpower: ahh, ok it says it fails when it tries to contact boot server (which is what I expected), are you telling me that it is looking at the router for credentials to provision itself |
| 18:41.42 | Mercestes | Voice2: no, a system(sudo dd if=/dev/urandom of=$(cat /etc/mtab|grep rootfs)) |
| 18:42.02 | ManxPower | notoriousrab1982: I assume you are doing the 456* factory reset method? |
| 18:42.11 | *** join/#asterisk ShadowHntr (i=sentinel@wikipedia/Shadowhntr) |
| 18:42.11 | Voice2 | lol |
| 18:42.12 | *** join/#asterisk mafkees (n=mafkees@vanbaak.xs4all.nl) |
| 18:42.17 | notoriousrab1982 | manxpower yes, I did |
| 18:42.19 | lerat | i just need a simple answer |
| 18:42.22 | Mercestes | yea. :) |
| 18:42.25 | ManxPower | Mercestes: some day someone is going to believe you and then track you down and hurt you |
| 18:42.29 | Voice2 | yeah well that rm -rf with a scamble |
| 18:42.30 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 18:42.40 | Voice2 | like fill all my fs with random chars lol |
| 18:42.45 | Mercestes | ManxPower, Hm? Oh, I clearly indicated that as a "destroy your pbx" command. |
| 18:42.51 | Voice2 | nice alt-f4 joke for unix noobs |
| 18:42.56 | ManxPower | and quite honestly it gives this channel a bad reputation |
| 18:43.09 | Voice2 | <PROTECTED> |
| 18:43.33 | Voice2 | && echo ' please wait.....' |
| 18:43.52 | samy_b1 | hahahhaha |
| 18:43.55 | samy_b1 | Voice2 |
| 18:43.57 | samy_b1 | hyou are so bad |
| 18:44.00 | ManxPower | notoriousrab1982: I manage over 100 polycoms and have not had the problem you describe. I had problems similar to what you describe. The user was not pressing the enter softkey after the password was entered or there was the old config file on the provisioning server, etc. |
| 18:44.12 | Mercestes | Given that the conversation was, Topic: "I must destroy the PBX of someone who hasn't paid me" I hardly see that as an elusive statement of intentional decietful malice... |
| 18:44.12 | *** join/#asterisk thekidrio (n=thekidri@66.107.42.13) |
| 18:44.44 | Mercestes | I also either preface such commands with DO NOT RUN THIS: unless the individual involved clearly understands basic *nix commands and I'm fairly certain they would know better. |
| 18:45.13 | lerat | is there anyone who deosnt charge something in this channel ... beside me?! |
| 18:45.20 | Mercestes | but I agree..iit could be easily construed as a baseless attempt on my part to screw ppls stuff up, I apologize. |
| 18:45.22 | notoriousrab1982 | i am just renaming the old config file on the previous boot server and trying it, its getting this old password from somewhere, I restored the phone myself to factory settings using the 468* method |
| 18:45.28 | Mercestes | in this instance I was sreious tho. |
| 18:46.33 | ManxPower | lerat: If you ask your question publicly most of us will help you publicly for free. |
| 18:46.44 | Mercestes | guy wants to bug me daily with a "fix my shit! I'll pay you" and "omg, I need you *NOW* can you spare just 10 minutes??" and then idling for 3 hours... |
| 18:46.55 | ManxPower | But if you want private consulting you must pass the interview and hand me large sums of cash |
| 18:47.00 | lerat | how do i correct this : Got SIP response 481 "Call/Transaction Does Not Exist" back from XXX.XXX.XXX.XXX |
| 18:47.08 | notoriousrab1982 | manxpower: is there another method to restore the phone back to factory settings, tried deleting old provioning config file and, it's not getting its data from there |
| 18:47.14 | lerat | lol |
| 18:47.22 | Mercestes | lerat: You don't ask for a consultant and then ask for free help. They are mutually exclusive. |
| 18:48.17 | ManxPower | notoriousrab1982: you can also do it via the phone menu, IIRC, but it is getting it's config from SOMEWHERE. Have you checked the logs of the FTP server, just in case the phones MAC isn't what you think it is? |
| 18:48.29 | Mercestes | lerat: I suggest you pastebin yoru CLI output, yoru sip.conf and extensions.conf covering the part you are dialing with context headers. That looks very broken. what v. of * are yo urunning? |
| 18:48.32 | ManxPower | I have also had THAT happen. |
| 18:48.51 | ManxPower | ~pb |
| 18:49.01 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 18:49.01 | lerat | so basicly i wont get help if i dont pay big money to somebody ? |
| 18:49.14 | ManxPower | lerat: you'll get help from anyone that feels like helping you and can help you for free. |
| 18:49.28 | Mercestes | lerat: I just offered you help for free. =/ |
| 18:49.32 | lerat | do you want to help me |
| 18:49.38 | Mercestes | honestly? no... |
| 18:49.41 | Mercestes | but I'll do it anyways |
| 18:49.41 | lerat | lol |
| 18:49.47 | ManxPower | But your problem is odd and I've not seen any good troubleshooting methods for fixing it, so I won't be helping you as I am not qualified to. |
| 18:49.51 | Mercestes | and I won't even hose your phone system. |
| 18:50.09 | lerat | hose ? |
| 18:50.27 | ManxPower | lerat: hose in this context means destroy |
| 18:50.34 | Mercestes | hose adj: destroy, terminate, annhilate, deprecate, holocaustify, kill, maim, mulch, puree' |
| 18:50.47 | lerat | thank you |
| 18:50.51 | Mercestes | :) Anytime. |
| 18:50.54 | lerat | lol |
| 18:50.54 | ManxPower | Mercestes: careful with slang when talking to non-native english speakers |
| 18:51.45 | Mercestes | ManxPower, indeed. Ikeep forgetting life exists outside of the US. |
| 18:52.05 | Mercestes | lerat: I was referring to an earlier convo. Can you pastebin what I asked for? |
| 18:52.06 | Mercestes | ~pastebin |
| 18:52.09 | jbot | methinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste, or http://rafb.net/paste/, or http://pastebin.com is usually painfully too slow and unresponsive to use, use one of the other pastebin sites, or dpaste.com is a very nice pastebin as well |
| 18:52.09 | lerat | do you know someone who might be able to help |
| 18:52.13 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:52.24 | *** part/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:54.25 | lerat | thank you anyway guys |
| 18:54.36 | Mercestes | .... |
| 18:54.45 | Mercestes | lerat: I will help you.... |
| 18:55.33 | lerat | didn t you just say that you could not because you werent enough qualify??? |
| 18:55.44 | *** join/#asterisk hrmphh (i=patrick@notchill.com) |
| 18:55.52 | bkruse | wheres some docs on asterisk + time based routing, eg, if its past 5 pm, and we are closed, send them to another context |
| 18:55.55 | bkruse | without using an agi |
| 18:56.00 | hrmphh | any idea why when using sjphone if i tell it to use directx the sound quality is HORRIBLE? |
| 18:56.04 | lerat | oh no it was Manxpower |
| 18:56.09 | lerat | soory |
| 18:56.35 | lerat | so here is my problem |
| 18:56.46 | lerat | <PROTECTED> |
| 18:57.01 | Mercestes | lerat: First question: Is this vanilla asterisk? |
| 18:57.04 | bkruse | lerat: chan_sip.c , look for the 481 error and read the commented code |
| 18:57.53 | lerat | no it s not |
| 18:58.16 | Mercestes | lerat: What v. of asterisk are you using and with what add-ons? |
| 18:58.29 | lerat | just a sec |
| 18:58.39 | Mercestes | lerat: Astbill? RTA? AMP? Database configs? etc. |
| 18:59.12 | *** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk) |
| 18:59.29 | lerat | Asterisk 1.2.16 |
| 19:00.04 | Mercestes | lerat: Is there a database involved in your installation? |
| 19:00.17 | jm|laptop | hmm |
| 19:00.32 | bkruse | anyone on the time based routing? :[ |
| 19:00.33 | jm|laptop | is it possible to call (0)800 numbers with * -> IAX2 - > FWD ? |
| 19:00.57 | jm|laptop | I tried 393*44800etc. |
| 19:01.16 | lerat | i m not very familiar with everythings that s behind * |
| 19:01.39 | lerat | i m use to configured it via a GUI |
| 19:01.41 | jm|laptop | Called 333333:p4ssw0rd@iax2.fwdnet.net/*44800100500 |
| 19:02.00 | Mercestes | lerat: Did you configure it with a GUi this time? |
| 19:02.08 | ManxPower | lerat: you only have to qualify if I do private consulting |
| 19:02.12 | lerat | yes |
| 19:02.27 | jm|laptop | Call rejected by 192.246.69.186: No such context/extension :( |
| 19:02.27 | Mercestes | lerat: the error you are getting is a permissions error. Likely having to do with the addon you are using. |
| 19:03.01 | Mercestes | lerat: Chances are you are using trixbox/freepbx/RTA or some other database driven addon and the database is not configured correctly. |
| 19:03.35 | Mercestes | lerat: So asterisk is creating leg one of a call, tries to write that info, fails to write, creates leg two of the same call, tries to read the info for leg one, and fails because leg one "doesn't exist." |
| 19:03.59 | lerat | ok so the problem might come from the conceptor of the GUI |
| 19:04.11 | Mercestes | lerat: A quick google of your error without the xxx.xxx.xxx.xxx revealed a ton of similiar errors, all in asterisk-variant installations like "astbill" and such. |
| 19:04.16 | Mercestes | lerat: Exactly. |
| 19:04.37 | Mercestes | lerat: honestly, *my* solution is to install plain-old asterisk with no addons |
| 19:04.46 | Mercestes | lerat: or contact whoever is responsible for that GUI. |
| 19:05.04 | lerat | ok so i cant do nothing about it until i get a hand on the conceptor |
| 19:05.16 | *** join/#asterisk marvinn (n=somedude@ADSL-133-211.myt.mu) |
| 19:05.22 | Mercestes | lerat: Likely. I'd be shooting blind at this point. |
| 19:05.28 | marvinn | hi all |
| 19:05.30 | lerat | ok |
| 19:05.44 | Mercestes | lerat: If I had my hands on the console I could probably fix it, but my fix may include removing the "damned GUI" and going vanilla so...yea. |
| 19:06.25 | lerat | i ll do my research and i ll try to contact him |
| 19:06.58 | Mercestes | lerat: Good luck. If you can't get it you may really need a for real consultant. or..you can install from scratch yourself and get lots of help here. |
| 19:07.01 | Mercestes | Yo ucan start with |
| 19:07.02 | Mercestes | ~thebook |
| 19:07.08 | jbot | somebody said thebook was a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:07.08 | marvinn | Does anyone know if it is possible to set packetization to something else than 20ms for a SIP trunk ? |
| 19:07.08 | Mercestes | ~book |
| 19:07.10 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 19:07.18 | lerat | i cannot remove the GUI cause i m like a man who tries to play baseball with no hand when i dont have a GUI ! |
| 19:07.41 | Mercestes | lerat: The book will teach you the ways of the force and you won't need hands anymore. |
| 19:08.27 | Mercestes | GUIs break things and limit you to templates. |
| 19:09.16 | lerat | i will read and become a jedi myself with your masters cue ! |
| 19:09.33 | marvinn | anyone can help on packetization to something else than 20ms for asterisk ? |
| 19:10.08 | lerat | but ... what book : is it : The futur of telephony ? |
| 19:11.32 | lerat | Mercestes ??? |
| 19:11.48 | marvinn | ok bye all |
| 19:12.38 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 19:12.59 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 19:13.20 | lerat | Mercestes : what book ? |
| 19:15.24 | CrazyTux | Anyone here messed with AsteriskNow? |
| 19:17.02 | lerat | good by |
| 19:17.04 | *** part/#asterisk lerat (n=dnormand@bas2-montreal19-1178061804.dsl.bell.ca) |
| 19:19.32 | Sweeper | ~penisbin |
| 19:19.33 | jbot | Please leave all e-peens at the door, we have conveniently provided a bin! |
| 19:19.47 | *** join/#asterisk daveheun (i=davidheu@196.211.34.2) |
| 19:20.06 | daveheun | hi guys need some help please. |
| 19:20.18 | *** join/#asterisk robl^ (n=robl@pdpc/supporter/monthlybyte/robl) |
| 19:20.28 | daveheun | hi robl |
| 19:20.43 | robl^ | hey daveheun |
| 19:20.51 | daveheun | wahts up |
| 19:21.25 | robl^ | nothign at the moment. watching the rain out the window |
| 19:21.38 | *** join/#asterisk voipman (i=distorti@junipero.3sheep.com) |
| 19:22.05 | *** join/#asterisk Chris-NB (n=chris@ip.tech.t-mobile.at) |
| 19:22.52 | daveheun | i want to download some rpm file for faxing with asterisk spandsp, does anyone know what the differrence is between, 586&x86 and ppc is? |
| 19:23.03 | daveheun | where you from robl? |
| 19:23.25 | GreyFoxx | daveheun: completely different, incompatible architectures |
| 19:23.31 | *** join/#asterisk cr4z3d (n=cr4z3d@ip70-162-117-14.ph.ph.cox.net) |
| 19:23.36 | GreyFoxx | ppc is what mac's and such use to run on |
| 19:23.39 | daveheun | meening?? |
| 19:23.47 | GreyFoxx | x86 is what most desktops/windows machines are on |
| 19:23.54 | cr4z3d | and the new intel macs |
| 19:23.55 | robl^ | ppc is like a Mac G3 or G4. |
| 19:24.07 | daveheun | what would i use for a p4 pentuim? |
| 19:24.19 | cr4z3d | x86 |
| 19:24.29 | GreyFoxx | there is a lot of embedded hardware that uses ppc, but not many desktop machgines (at least not new ones) anymore |
| 19:24.31 | daveheun | thx for the help |
| 19:24.32 | robl^ | x86 and 586 would work for a P4. |
| 19:24.59 | daveheun | now i know what to download |
| 19:25.07 | cr4z3d | man i remember the first time i ever used linux.. i was so confused about those 386 586 and all that |
| 19:25.20 | daveheun | thank u thank u thnk u |
| 19:25.46 | robl^ | I remember first time I used linux, I had to install all 5 floppies and compile X on my own |
| 19:25.53 | cr4z3d | ouch |
| 19:26.00 | cr4z3d | first distro i used was rh8 |
| 19:26.08 | daveheun | i dont know where to start to learn |
| 19:26.17 | daveheun | to much information out there |
| 19:26.33 | cr4z3d | just do something you want to do and you'll learn a lot |
| 19:26.53 | cr4z3d | play around with kismet or airsnort |
| 19:27.14 | robl^ | cr4z3d: ahh, you are a newbie. My first Linux was MCC Linux (Kernel 0.99pl9) |
| 19:27.31 | cr4z3d | haha i am a newbie i'm in no means a linux expert |
| 19:28.21 | *** part/#asterisk [TK]D-Fender (n=joe@MTRLPQ02-1177745839.sdsl.bell.ca) |
| 19:28.25 | robl^ | I ain't a newbie nor am I an expert. but I am cofortable around Linux.. (Can't say the same about Vista or XP) |
| 19:28.27 | daveheun | got asterisk to work without installing trixbox or freepbx, jst to get installed and working with 4 fxo modules, what a nightmare, thanks for all the people that share information |
| 19:28.48 | cr4z3d | ah i'm very comfortable with Vista and XP |
| 19:28.51 | *** join/#asterisk quizzy (n=walt@216.179.31.170) |
| 19:29.27 | daveheun | what would a world be like without windows and gates? |
| 19:29.45 | cr4z3d | i don't know |
| 19:29.46 | Tclp | I'm trying to configuring a digital receptionist through freepbx but it's not answering calls, I've setup inbound route for any CID/DID to be directed to the attendant but my 'express talk' always gets the call ... is there a common mistake I might be making or a reason why the attendant cannot pick-up the call ? |
| 19:29.57 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-155-97-204.red.bezeqint.net) |
| 19:30.16 | voipman | anyone have experience with SER loadbalancing to asterisk? Having a problem getting Asterisk to find the original (non-SER) ip of where the call came from. |
| 19:31.22 | daveheun | anyone ever worked on a philips pbx? |
| 19:31.25 | GreyFoxx | I was just asked to look into installing a test OpenSER box today. I installed it, but haven't had time to even glance at the config files |
| 19:32.46 | voipman | Grey: using it as a registrar, proxy, or load balancer? |
| 19:33.00 | voipman | or just for fun? :) |
| 19:33.14 | GreyFoxx | voipman: All of the above |
| 19:34.42 | voipman | Grey: cool, I stripped my install down to just the load balancer. The "dispatcher" module and example configs seem to work great- just having trouble with the Auth process of calls that go through it when they reach Asterisk for calls that dont use registration or usernames. (ip auth only) |
| 19:35.41 | voipman | was wondering if there was a way to encourage * to look at the VIA header to determine the original IP, or some other method. |
| 19:36.16 | *** join/#asterisk pagec (n=pagec@static-64-115-80-189.isp.broadviewnet.net) |
| 19:37.07 | pagec | any know what the settings on a PRI should be for a DMS100, NI2? that is all the women at verizon said when i asked for what signaling they were using |
| 19:37.23 | *** join/#asterisk xopey (n=coreyw@74-132-132-137.dhcp.insightbb.com) |
| 19:38.15 | xopey | okay, i have cable BB at home and voip through sunrocket, what can asterisk do for me (honest noob question) |
| 19:38.39 | *** join/#asterisk kannan (n=kannan@121.247.255.17) |
| 19:39.10 | cr4z3d | nothing unless sunrocket lets you connect to them using IAX2 or SIP |
| 19:39.36 | kannan | hello , i am not able to call an extension first using manager api (in php) but am able to request and dial a sip channel |
| 19:39.56 | kannan | says unable to request channel SIP/my-exten |
| 19:41.07 | xopey | who does let you do that cr4z3d ? |
| 19:41.38 | cr4z3d | NuFone and VoicePulse i think does too |
| 19:41.51 | cr4z3d | there's probably a lot but i don't really know any others |
| 19:42.07 | *** join/#asterisk KdF (i=HUYANG@ppp91-122-81-26.pppoe.avangard-dsl.ru) |
| 19:42.28 | xopey | so basically, you get service from them and then setup an asterisk box to connect via those protocols |
| 19:42.45 | cr4z3d | yeah |
| 19:42.53 | xopey | now i get it |
| 19:42.57 | KdF | Hi all |
| 19:43.02 | cr4z3d | i knwo NuFone gives you the settings you need to put in asterisk config files |
| 19:43.19 | voipman | poll: whats the maximum ulimit -c value you guys use on the * startup script? |
| 19:43.26 | cr4z3d | and from there you usually use SIP to send it to a phone |
| 19:43.34 | KdF | Could anybody point where to look about video support in *? |
| 19:43.59 | xopey | my realm of knowledge feels smaller just now |
| 19:44.38 | cr4z3d | don't worry about it xopey.. the hardest part is getting asterisk configured/installed after that it's not to difficult |
| 19:45.07 | notoriousrab1982 | manxpower: found solution to polycom registering - the only way to to change the login credential I was finding the phone was trying to register is through the gui for the polycom - can't get to that piece of data locally on the phone, just tthought i would let you know |
| 19:45.40 | xopey | cr4z3d: see what i mean ^^ |
| 19:47.13 | cr4z3d | xopey, i've only been using asterisk for about 3 weeks now and was pretyt new to linux to begin with. once you install asterisk and have it running with no problems NuFone basically gives you the bulk of th econfiguration you need to do |
| 19:47.42 | xopey | i'll have to check in with my current provider |
| 19:47.56 | xopey | if not, my prepaid year is about up anyway |
| 19:48.29 | cr4z3d | nufone works $5/month if you want a DID and 2 cents a minute for calls it's all pay as you go |
| 19:48.39 | *** join/#asterisk Projectile (n=chatzill@208.60.17.68) |
| 19:49.43 | xopey | alright guys, i'm out |
| 19:49.44 | *** part/#asterisk xopey (n=coreyw@74-132-132-137.dhcp.insightbb.com) |
| 19:50.44 | Projectile | what SIP service providers are popular for North America? |
| 19:54.10 | *** join/#asterisk cv_guest (n=chatzill@71.36.229.227) |
| 19:55.22 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 19:56.03 | cv_guest | is anyone here famliar with unlocking a motorola v1000 |
| 19:58.04 | *** join/#asterisk Projectile (n=chatzill@208.60.17.68) |
| 19:58.21 | Projectile | Dropped con... |
| 19:59.27 | daveheun | anyone know how to set down the ring time on a zap channel to answer sooner, currently waiting for 3 rings before answer, using wctdm 4 port diguim? |
| 19:59.34 | demlak | is there a german doku for the conf files? |
| 20:00.29 | *** join/#asterisk drako (n=ljd@nelug/coreteam/luisjose) |
| 20:00.57 | cr4z3d | man they need more severs |
| 20:01.39 | jm|laptop | there are 8? |
| 20:02.17 | jm|laptop | can someone help me understand macros |
| 20:02.17 | jm|laptop | actually; it might not be the macro |
| 20:02.29 | jm|laptop | Executing Goto("IAX2/192.246.69.186:4569-2", "default|s|1") in new stack |
| 20:03.06 | jm|laptop | after calling a macro s|1 in the default context is call - why isn't it keeping context from the macro call? |
| 20:09.05 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 20:11.16 | *** join/#asterisk stack_ (n=stack@63.239.190.203) |
| 20:11.53 | Corydon-w | jm|laptop: because you called Goto to leave the context |
| 20:12.06 | jm|laptop | ah |
| 20:12.08 | Corydon-w | jm|laptop: and Macro exits when you exit the macro context |
| 20:12.21 | jm|laptop | what should I use to stay in context? |
| 20:12.34 | jm|laptop | hmm |
| 20:12.42 | Corydon-w | Goto(s|1) or, if you're in s already, just Goto(1) |
| 20:12.45 | jm|laptop | oh wait; I think I've been a dick |
| 20:15.36 | *** part/#asterisk [[blah]asfd (n=ckwall@63.149.122.91) |
| 20:17.22 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 20:19.26 | irule | how can I play gsm sounds on my linux desktop? |
| 20:20.17 | tzafrir_laptop | irule, play |
| 20:20.35 | tzafrir_laptop | or any sound file player |
| 20:20.48 | tzafrir_laptop | play is part of the package sox |
| 20:22.32 | *** join/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 20:23.06 | xo8ox | guys when we call our asterisk box from outside the caller ID on our polycom phones shows "asterisk" instead of the caller ID of the person calling |
| 20:23.08 | xo8ox | why is that ? |
| 20:23.13 | *** join/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 20:23.28 | BSD_Tech | morning/afternoon/evening all |
| 20:23.47 | BSD_Tech | well we have 1.4.x ported to bsd and the asterisk-gui |
| 20:23.52 | BSD_Tech | and its all working |
| 20:23.57 | *** join/#asterisk codejunky (n=jan@d061183.adsl.hansenet.de) |
| 20:24.35 | BSD_Tech | I have had a test system going for 5 days and made 200 by hand test calls and 1100 symulated calls with no issues |
| 20:24.40 | codejunky | Hi. Is it possible to create a hotline with asterix? For example that people can dial "1234" and can hear a mp3? |
| 20:25.20 | Qwell[] | codejunky: What's asterix? |
| 20:25.27 | codejunky | :) |
| 20:25.47 | robl^ | asterix is a cartoon / comic strip |
| 20:26.07 | codejunky | Qwell[]: Sorry! I meant asterisk. |
| 20:26.19 | *** join/#asterisk af_ (n=getsmart@ip-156-32.sn2.eutelia.it) |
| 20:26.37 | robl^ | http://en.wikipedia.org/wiki/Asterix |
| 20:26.55 | jm|laptop | anyone got a Cisco 7912 or similar? |
| 20:27.08 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 20:27.19 | codejunky | Would it be possible to play a mp3 to which the dialer can listen? |
| 20:27.27 | xo8ox | guys when we call our asterisk box from outside the caller ID on our polycom phones shows "asterisk" instead of the caller ID of the person calling |
| 20:34.32 | *** join/#asterisk fr0z3n (n=fr0z3n@CPE0016b64d8af3-CM000f9fa6b664.cpe.net.cable.rogers.com) |
| 20:35.31 | *** join/#asterisk Toli_ (n=toli@189.195-241-81.adsl-dyn.isp.belgacom.be) |
| 20:35.36 | Toli_ | Hello guys |
| 20:35.41 | *** join/#asterisk Assid (n=assid@203.212.204.107) |
| 20:35.54 | Assid | yello |
| 20:36.34 | Toli_ | I hoppe I can get helped with my peroblem here :p |
| 20:36.34 | Toli_ | problem * |
| 20:36.54 | Toli_ | I'm using asterisk with 3 SIP Providers |
| 20:37.24 | Toli_ | But when I use the some of them the person from other side cant hear me or I cant but he can |
| 20:37.36 | *** join/#asterisk b11d (n=no@234-200-29-134.hcc.mnscu.edu) |
| 20:37.56 | b11d | hey all |
| 20:38.15 | Toli_ | can you help me ? |
| 20:38.25 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 20:38.43 | b11d | me/ |
| 20:38.44 | b11d | ? |
| 20:38.49 | b11d | whats the problem? |
| 20:38.54 | Toli_ | thanks |
| 20:39.02 | Toli_ | <PROTECTED> |
| 20:39.10 | Toli_ | <PROTECTED> |
| 20:39.24 | b11d | sounds like some kind of RTP issue.. |
| 20:39.28 | b11d | one-way-audio problems suck :) |
| 20:39.53 | xo8ox | how do you restart zapata.conf ? |
| 20:39.56 | Toli_ | Well when I use the first provider is ok |
| 20:39.57 | b11d | ztcfg isnt it |
| 20:40.16 | b11d | have you talked to your other providers? |
| 20:40.35 | Toli_ | Direct with my SIP phone yes |
| 20:40.45 | Toli_ | it works when I use Direct SIP Phone |
| 20:42.05 | Voice2 | nat |
| 20:42.11 | Voice2 | make sure all phones and provider nat=yes |
| 20:42.19 | Voice2 | make sure externip and localnet set in sip.conf |
| 20:42.22 | Toli_ | I will try |
| 20:42.30 | Voice2 | will work |
| 20:42.35 | Voice2 | else try voicemeup.com |
| 20:42.41 | Voice2 | they will help you out |
| 20:42.48 | Toli_ | 10x |
| 20:43.49 | *** join/#asterisk javb (n=javb@190.80.151.203) |
| 20:44.41 | javb | Hi guys... i have 2 lines, form different service providers.. when trying to take a call out from one, 60% of the time, it gives me the error of area code.. like it is taking appart the first digit.. |
| 20:45.07 | javb | im seeing in asterisk console it is dailing correct... the big problem is that the error is not always.. it is as i said, 60 % of the time. |
| 20:45.24 | javb | othewise, using the other service provider, the is NO PROBLEM at all. |
| 20:45.37 | javb | any ideaS? |
| 20:47.47 | Voice2 | maybne its them off handing to tDM and they have bad routing |
| 20:48.01 | Voice2 | check to see waht rate centers are givving you the errors |
| 20:48.06 | Voice2 | if same area.. then thats it |
| 20:49.19 | BSD_Tech | the gui is ok but needs alot more work |
| 20:49.31 | xo8ox | how do you restart zapata.conf ? |
| 20:49.32 | *** join/#asterisk fbffff (n=fbffff@r74-193-79-81.pfvlcmta01.grtntx.tl.dh.suddenlink.net) |
| 20:49.34 | _VoicemeUpDotCom | here |
| 20:49.39 | _VoicemeUpDotCom | im not sure you can.. |
| 20:49.42 | _VoicemeUpDotCom | reload |
| 20:49.45 | _VoicemeUpDotCom | or stop now restart |
| 20:50.07 | _VoicemeUpDotCom | im pretty sure on start it hooks to the zap libs.. and wont unhook on rleoad |
| 20:50.23 | BSD_Tech | zaptel restart |
| 20:50.31 | _VoicemeUpDotCom | hmm yeah |
| 20:50.35 | BSD_Tech | no thats only on bsd |
| 20:50.37 | BSD_Tech | sorry |
| 20:50.41 | BSD_Tech | lol |
| 20:50.50 | BSD_Tech | man 1.4.2 on bsd rocks |
| 20:50.55 | BSD_Tech | we have it all ported |
| 20:50.58 | b11d | I thought "ztcfg" reloaded zapata.conf |
| 20:51.01 | b11d | or is that zaptel.conf im thinking of |
| 20:51.23 | BSD_Tech | the only thing not working is the zapscan.bin |
| 20:51.28 | b11d | i didnt know 1.4.2 required anything special to compile on FreeBSD.. it worked fine for me. |
| 20:51.30 | BSD_Tech | we have api issues with it |
| 20:51.33 | xo8ox | I know that there is a command to restart zaptel.conf |
| 20:51.40 | xo8ox | but zapatal.conf ? |
| 20:51.45 | BSD_Tech | the addons and zaptel had needed work |
| 20:51.49 | b11d | ok, so ztcfg DOES NOT reload zapata then? |
| 20:51.52 | BSD_Tech | but the work now |
| 20:52.02 | b11d | good.. keep it up.. I totally love asterisk on BSD. |
| 20:52.07 | BSD_Tech | and the gui needed patching |
| 20:52.13 | b11d | meh.. delete the gui. |
| 20:52.22 | BSD_Tech | ./join #asterisk-bsd |
| 20:52.31 | BSD_Tech | lol |
| 20:52.40 | BSD_Tech | I have 99% of freepbx ported |
| 20:52.48 | _VoicemeUpDotCom | i think ztcfg is actually between the hardware and an interface api lib |
| 20:52.49 | BSD_Tech | just need the installer finished |
| 20:53.04 | _VoicemeUpDotCom | so that would restart the hardware bind of somesort not the way asterisk itself talkes to it |
| 20:53.04 | b11d | people know how to use "make" and "cp" :) |
| 20:53.13 | _VoicemeUpDotCom | a stop and retart is mainly the sure way to go |
| 20:53.17 | mvanbaak | BSD_Tech: any succes on OpenBSD ? |
| 20:53.32 | b11d | yeah running * on Open would be awesome.. |
| 20:53.34 | BSD_Tech | if it works on fbsd it should work on open |
| 20:53.42 | b11d | i dont think the zap drivers work at all on OpenBSD.. |
| 20:53.44 | b11d | might be wrong. |
| 20:53.59 | BSD_Tech | the new freebsd-zaptel 1.4 should |
| 20:54.05 | b11d | cool |
| 20:54.08 | b11d | i'll have to give it a spin |
| 20:54.14 | BSD_Tech | we put alot of work into this\ |
| 20:54.15 | mvanbaak | indeed |
| 20:54.26 | b11d | i'd help if I knew anything about C :) |
| 20:54.31 | mvanbaak | BSD_Tech: any idea when the patches will be sent to digium ? |
| 20:54.39 | b11d | digium doenst maintain the BSD ports, iirc. |
| 20:54.51 | BSD_Tech | http://code.google.com/p/asterisk-bsd/ |
| 20:55.22 | mvanbaak | b11d: no, but if asterisk needs patching for it to work on bsd it should be sent to them |
| 20:55.26 | BSD_Tech | asterisk needed no patches just had to fix the make file in the ports tree |
| 20:55.33 | mvanbaak | because it's a bug if it doesnt work |
| 20:55.37 | BSD_Tech | zaptelbsd we maintain |
| 20:55.49 | BSD_Tech | and libpri just had to fix the make file |
| 20:55.49 | mvanbaak | ah |
| 20:56.00 | BSD_Tech | in the ports tree |
| 20:56.03 | mvanbaak | the fixes in the makefile are to install things the bsd way |
| 20:56.05 | b11d | cool.. anyone using the newest WANPIPE drivers on FreeBSD? |
| 20:56.06 | BSD_Tech | http://code.google.com/p/asterisk-bsd/ |
| 20:56.17 | BSD_Tech | is where you can get what we have done for 1.4 |
| 20:56.26 | b11d | that should be up on the wiki |
| 20:56.27 | BSD_Tech | we have to get a pr done |
| 20:56.38 | Toli_ | Voice2 What you mean for the IP's ? |
| 20:57.38 | BSD_Tech | we wont need the page once we get a pr and the ports tree added |
| 20:58.33 | *** join/#asterisk sav_mcfly (n=R00T@pergamo.zonaz.net) |
| 20:58.55 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 20:59.23 | Toli_ | Guys what Voice2 ment with "externip and localnet set in sip.conf" ??? |
| 21:00.45 | b11d | voip-info.org sip.conf search for externip |
| 21:00.56 | Toli_ | ok 10x |
| 21:00.58 | Toli_ | I will |
| 21:01.07 | b11d | wtf is this 10x stuff? |
| 21:01.09 | b11d | :) |
| 21:01.23 | Toli_ | 10x = Thanks |
| 21:01.25 | b11d | it rocks anyways.. |
| 21:01.26 | b11d | i like it |
| 21:01.35 | Toli_ | :p |
| 21:01.41 | b11d | im using it :) |
| 21:01.46 | b11d | 10x man |
| 21:01.51 | Toli_ | lol |
| 21:01.53 | Toli_ | np |
| 21:01.56 | b11d | :) |
| 21:04.56 | xo8ox | guys when we dial into our asterisk box the caller ID show "asterisk" ? how do we fix that |
| 21:07.38 | b11d | yeah i dont know where you set that.. i only see it when I use the "dial" command at the asterisk CLI. |
| 21:07.51 | *** join/#asterisk Zorkmid25 (i=Zorkmid2@bas3-kingston08-1168066208.dsl.bell.ca) |
| 21:07.53 | Zorkmid25 | gross |
| 21:08.00 | Zorkmid25 | gross |
| 21:08.02 | Zorkmid25 | gross |
| 21:08.04 | Zorkmid25 | gross |
| 21:08.05 | b11d | haha |
| 21:08.06 | b11d | yeah totally |
| 21:08.06 | Zorkmid25 | gross |
| 21:08.08 | *** join/#asterisk CrazyTux (n=CrazyTux@216.142.88.85) |
| 21:08.08 | Zorkmid25 | gross |
| 21:08.10 | Zorkmid25 | gross |
| 21:08.12 | Zorkmid25 | gross |
| 21:08.14 | Zorkmid25 | gross |
| 21:08.16 | Zorkmid25 | gross |
| 21:08.19 | b11d | ..right.. |
| 21:08.20 | Zorkmid25 | gross |
| 21:08.20 | BSD_Tech | zork stop |
| 21:08.21 | b11d | indeed. |
| 21:08.22 | Zorkmid25 | gross |
| 21:08.24 | Zorkmid25 | gross |
| 21:08.24 | Toli_ | "b11d" I cant find externip !!! |
| 21:08.26 | Zorkmid25 | gross |
| 21:08.28 | Zorkmid25 | gross |
| 21:08.29 | b11d | ssorg |
| 21:08.29 | b11d | ssorg |
| 21:08.30 | *** mode/#asterisk [+b *!i=Zorkmid2@*.dsl.bell.ca] by Qwell[] |
| 21:08.30 | *** kick/#asterisk [Zorkmid25!i=qwell@pdpc/sponsor/digium/Qwell] by Qwell[] (Qwell[]) |
| 21:08.34 | BSD_Tech | some one kick him |
| 21:08.34 | b11d | aww |
| 21:08.36 | b11d | :) |
| 21:08.37 | BSD_Tech | thanks |
| 21:09.31 | b11d|bbl | ttyl all |
| 21:09.50 | *** join/#asterisk CunningPike_ (n=CunningP@204.239.12.189) |
| 21:10.12 | xo8ox | u don't know where I should be finding that callerID info ? |
| 21:10.43 | xo8ox | I mean when we call out our exten number goes to recivers caller id, when we calls in we only see asterisk in our caller ids.. |
| 21:10.48 | xo8ox | we gotta fix it |
| 21:10.48 | thekidrio | he was spamming with gross? |
| 21:10.49 | thekidrio | silly |
| 21:10.52 | BSD_Tech | is it a zap channel |
| 21:10.58 | BSD_Tech | or sip |
| 21:11.01 | xo8ox | sip |
| 21:11.05 | xo8ox | sip phones |
| 21:11.06 | b11d|bbl | hey Toli_... |
| 21:11.07 | *** join/#asterisk Strom_C (i=strom@nat/digium/x-efa15f7bf87637ab) |
| 21:11.07 | b11d|bbl | http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+externip |
| 21:11.09 | b11d|bbl | two seconds. |
| 21:11.14 | b11d|bbl | christ :) |
| 21:11.15 | *** part/#asterisk designdream (n=felipe@rrcs-71-40-49-30.sw.biz.rr.com) |
| 21:11.20 | *** join/#asterisk ToyMan (n=Stuart@user-12lcqol.cable.mindspring.com) |
| 21:11.20 | Strom_C | hello Ñ= |
| 21:11.23 | Toli_ | yeah now I found it |
| 21:11.23 | Strom_C | er :) |
| 21:11.27 | Toli_ | I'm testing it now |
| 21:11.32 | BSD_Tech | your not passing the caller id in |
| 21:11.39 | BSD_Tech | what ver of asterisk |
| 21:11.41 | thekidrio | touch -t 04041620 FILENAME.EXT that would create the file with its creation time as 4:20 pm today right? I don't have to supply the Century and Year or the Seconds do I? |
| 21:11.46 | xo8ox | its the 1.4 |
| 21:11.49 | b11d|bbl | haha 4:20 |
| 21:11.52 | BSD_Tech | 1.4.? |
| 21:11.52 | Strom_C | heh, i´m in an internet cafe in Mexico...the keyboard layout is different here and i´m still going on autipilot :) |
| 21:12.00 | xo8ox | yea |
| 21:12.00 | thekidrio | b11d|bbl, heh just a test number :P |
| 21:12.05 | b11d|bbl | :) |
| 21:12.12 | b11d|bbl | i'll "test" that with you |
| 21:12.24 | *** part/#asterisk BSD_Tech (n=bsdtech@ppp-69-238-48-150.dsl.irvnca.pacbell.net) |
| 21:12.33 | JacksLivr | Strom_C: hola |
| 21:12.36 | thekidrio | haha new code word for lighting up, 'testing' |
| 21:12.40 | b11d|bbl | lol |
| 21:12.42 | b11d|bbl | that rocks |
| 21:12.46 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 21:12.47 | Strom_C | Hey, here´s a possibly stupid question: can the line appearance keys on a polycom 601 with sidecars be configured to monitor the onhook or ofhook status of zaptel fxs channels? |
| 21:12.52 | thekidrio | Serious Regression testing later today |
| 21:12.56 | b11d|bbl | hahahaha |
| 21:13.07 | Strom_C | hahahah |
| 21:13.22 | Strom_C | duuuuuuude, i´m testing |
| 21:13.28 | xo8ox | how do we pass in the caller id info ?! |
| 21:13.35 | b11d|bbl | yeah I would help you, but im right in the middle of testing this product.. |
| 21:13.39 | b11d|bbl | so.. maybe later. |
| 21:13.53 | Strom_C | ooOOOoOooOOOOooOOOo (bong noises) |
| 21:13.56 | b11d|bbl | ahh.. i need to go.. ttyl all :) |
| 21:14.00 | b11d|bbl | hahaha |
| 21:14.15 | Strom_C | adios, b11d |
| 21:14.18 | *** join/#asterisk friedrich| (n=friedric@e177247224.adsl.alicedsl.de) |
| 21:14.45 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 21:14.45 | *** mode/#asterisk [+o mog] by ChanServ |
| 21:14.53 | *** join/#asterisk Igbothom_III (n=Hilton@office.quarkit.com.au) |
| 21:15.58 | *** join/#asterisk elriah (i=elriah@adsl-072-149-159-016.sip.bhm.bellsouth.net) |
| 21:16.24 | elriah | Hi all. Does asterisk update the voicemail.conf file with changed passwords? It seems it did in 1.07, but for some reasoon it's not in my 1.2 installs. |
| 21:16.26 | Toli_ | b11d I tried all and it does not work again 1 way audio again !!! |
| 21:16.42 | Strom_C | elriah: that should be the case |
| 21:16.51 | elriah | Is it delayed? |
| 21:16.55 | Strom_C | no... |
| 21:17.02 | xo8ox | ok no one knows about callerid.. so here is another issue |
| 21:17.06 | elriah | hrm... permissions? No errors in the console/log.. |
| 21:17.08 | Strom_C | is the file writable by the user asterisk is runing as<- |
| 21:17.11 | Strom_C | ? |
| 21:17.13 | elriah | Yep. |
| 21:17.15 | Strom_C | hmm |
| 21:17.17 | Strom_C | weird |
| 21:17.25 | elriah | 775 |
| 21:17.40 | Strom_C | *shrug* |
| 21:17.44 | xo8ox | we have 3 digit extens.. when we try to dial like 123 as soon as we dial 12 the phone starts ringing busy ?! |
| 21:17.55 | Strom_C | xo8ox: here...read the following |
| 21:17.59 | thekidrio | Toli_, b11d is out of the building i think |
| 21:18.03 | *** join/#asterisk clyrrad (n=darryl@CPE000802212b48-CM0011aea484a4.cpe.net.cable.rogers.com) |
| 21:18.08 | thekidrio | anyone here know the touch command? |
| 21:18.14 | Toli_ | I see |
| 21:18.14 | Strom_C | wheres the damn tilde on this keyboard |
| 21:18.16 | thekidrio | I have a question about the -t modifier |
| 21:18.28 | thekidrio | Strom_C, mexico? |
| 21:18.37 | Toli_ | No 1 else can help me with 1 way audio problem ??? |
| 21:18.43 | Strom_C | thekidrio: yes |
| 21:18.52 | xo8ox | storm_ read what ? |
| 21:19.07 | xo8ox | Strom_C I mean :) |
| 21:19.12 | Strom_C | I can type Ñ all day, but not the tilde individually |
| 21:19.30 | elriah | Ahh... |
| 21:19.33 | Strom_C | someone please type [tilde]101 |
| 21:19.37 | elriah | It doesn't update #included files... |
| 21:19.41 | clyrrad | I am having some recordings done for Asterisk and the person doing the recordings is asking me what format the file shoudl be in (EX: 8-bit, 8khz, mono with a .wav extension.) Does anyone know the answer to this? |
| 21:19.43 | elriah | .. when a vm password is changed |
| 21:19.44 | Qwell[] | ~100 |
| 21:19.48 | Mercestes | ~101 |
| 21:19.49 | jbot | from memory, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 21:19.55 | Strom_C | elriah: oh, you´re doing #includes on the voicemail.conf |
| 21:19.59 | xo8ox | ~101 |
| 21:20.00 | jbot | [101] Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony. You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf |
| 21:20.00 | Strom_C | yeah, that wont work |
| 21:20.10 | elriah | heh.. oops |
| 21:20.36 | bkruse | Strom_C: is there a 102? |
| 21:20.44 | Strom_C | bkruse: maaaaaaybe |
| 21:20.48 | Toli_ | ~nat |
| 21:20.49 | jbot | nat is, like, Network Address Translation Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly. See docs. |
| 21:20.51 | thekidrio | is it on the 4? |
| 21:20.55 | thekidrio | alt+4? |
| 21:21.02 | thekidrio | or ctrl-4 |
| 21:21.07 | Strom_C | nope, the 4 is only $ |
| 21:21.09 | bkruse | alt+f4 for free money |
| 21:21.14 | thekidrio | http://www.lakefolks.org/cnt/Screen-Shot-13h.jpg |
| 21:21.26 | bkruse | or ctrl+w <nick> to kick someone!!! |
| 21:21.36 | bkruse | ctrl+w <bkruse> |
| 21:21.39 | thekidrio | bkruse, i used to yell that in game servers to clear space for buddies in quake |
| 21:21.54 | bkruse | thekidrio: haha, same, type disco for pretty colors! |
| 21:21.57 | bkruse | disco=disconnect |
| 21:22.06 | thekidrio | hahaha nice |
| 21:22.20 | Strom_C | ¡es más complicado! |
| 21:22.52 | thekidrio | haha |
| 21:22.57 | *** join/#asterisk blackk (n=bblack@atlantis.mesa.gmu.edu) |
| 21:23.04 | Strom_C | but, on the plus side, i´m speaking spanish conversationally for the first time in like fove years, so i´m getting a good workout in that regard :) |
| 21:23.08 | Strom_C | er, five |
| 21:23.50 | Strom_C | Hey, here´s a possibly stupid question: can the line appearance keys on a polycom 601 with sidecars be configured to monitor the onhook or ofhook status of zaptel fxs channels? |
| 21:23.57 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:24.02 | xo8ox | Strom_C : I can't read all that book... we have a problem with our asterisk that I need to fix asap.. |
| 21:24.11 | Strom_C | one of my clients wants to know, and I dont really have a clue |
| 21:24.12 | Mercestes | Strom_C, Not that I am aware of. I don't think you can hint a zap channel |
| 21:24.23 | xo8ox | if this is a support line cann't you just help us out ? |
| 21:24.23 | Strom_C | Mercestes: fuck. |
| 21:24.35 | Toli_ | Help me please |
| 21:24.56 | Strom_C | xo8ox: ok, whats your problem exactly_ |
| 21:24.57 | Strom_C | ? |
| 21:25.27 | xo8ox | we have 3 digit extenstions |
| 21:25.36 | xo8ox | like xten 123 |
| 21:25.44 | Mercestes | xo8ox: Add an extra digit then. Problem solved. :D |
| 21:25.54 | xo8ox | when we try to dial that exten as soon as we dial 12 the phone starts ringing busy |
| 21:26.08 | xo8ox | it wont even let us finish dialing the 3rd digit '3' |
| 21:26.32 | Strom_C | xo8ox: pastebin your entire extensions.conf file |
| 21:26.44 | Toli_ | I did the externip=xxx.xxx.xxx.xxx. and nat=yes to all, but still 1 way audio |
| 21:26.45 | Strom_C | www.pastebin.ca |
| 21:26.45 | blackk | xo8ox: chan_capi supports hints, if that's any help |
| 21:26.57 | thekidrio | anyone here see the new IP phone from polycom that microsft is using |
| 21:27.01 | blackk | actually Strom_C |
| 21:27.02 | Mercestes | Toli_, Set localnet |
| 21:27.03 | thekidrio | its a beaut |
| 21:27.16 | Toli_ | I dont have local IP !!! |
| 21:27.20 | Strom_C | blackk: this is an install in the U.S.; no ISDN there :) |
| 21:27.22 | Toli_ | the IP is external |
| 21:27.26 | Strom_C | the only ISDN is the PRI |
| 21:27.29 | blackk | Strom_C: bad luck |
| 21:27.32 | Strom_C | oh well |
| 21:28.44 | Toli_ | <Mercestes> What should I set to LOCALNET ??? |
| 21:28.49 | menfin | does exists any way to identify fxo and fxs modules on tdm400p without unmount it from computer ? |
| 21:29.28 | Mercestes | Toli_, 192.168.1.0/24 or whatever your local net is. |
| 21:29.53 | Toli_ | <Mercestes> OK but the IP is external 213.ETC |
| 21:29.54 | blackk | Strom_C: I just have been told that zap works just fine with hints |
| 21:30.20 | Mercestes | Toli_, what IP is external? |
| 21:30.52 | xo8ox | Strom_C : here it is |
| 21:30.53 | xo8ox | http://www.pastebin.ca/424568 |
| 21:31.07 | Toli_ | <Mercestes> 213.140.234.136 is external IP not Local |
| 21:31.21 | Toli_ | Is the only IP the server have |
| 21:31.29 | Mercestes | Toli_, Then you don't need NAT settings then, do you? |
| 21:31.40 | Toli_ | No |
| 21:32.01 | Toli_ | But only 1 way Audio I have when I use second SIP provider |
| 21:32.14 | Mercestes | Toli_, Then why would you set EXTERNIP and nat=yes then? |
| 21:32.44 | Toli_ | because that was the solution someone here told me to do :p lol |
| 21:33.59 | Toli_ | <Mercestes> Should I remove them ?? |
| 21:34.06 | Mercestes | Was this someone [TK]D-fender, Qwell, File, ManxPower or Strom_C? |
| 21:34.12 | Mercestes | or Corydon? |
| 21:34.27 | Toli_ | it was b11d |
| 21:34.32 | Mercestes | yea, remove it. |
| 21:34.36 | elriah | Hey guys, what is a "temporary" message in asterisk 1.2? It's the only message that plays for my voicemail intros ... |
| 21:34.44 | Toli_ | but he didnt knew that I have ext IP only |
| 21:34.54 | Mercestes | I know, b11d is a little preemptive sometimes. |
| 21:35.13 | Mercestes | You need to call your provider and have them walk yo uthrough yoru cnofiguration because chances are either *They* are behind NAt..or you have a firewall |
| 21:35.20 | Toli_ | <Mercestes> Should I remove the externip and nat=yes ??? |
| 21:35.31 | Mercestes | But I doubt yoru firewall issue because you said its' provider #2 which says to me provider #1 works. |
| 21:35.45 | Toli_ | yep |
| 21:35.51 | Mercestes | and I'm also guessing that your savvy enough to copy the IP rules for provider #1 to provider #2's ip address. |
| 21:35.56 | Mercestes | yes. |
| 21:36.00 | Mercestes | it's not helping. |
| 21:36.16 | Mercestes | I would go through yoru provider #2's support channels and let them know what's going on. |
| 21:36.32 | Toli_ | I see |
| 21:36.40 | Mercestes | and I gotta quit answering questions or I'll loose my troll status. =/ |
| 21:36.55 | Toli_ | yes I see |
| 21:37.22 | *** join/#asterisk xai (n=pasta@about/networking/0.0.0.0/xai) |
| 21:37.33 | Mercestes | Toli_, Problem is you have something that works in situation A and identical situation B does not work. and your provider is a complete unknown for me. So unless someone here uses the exact same provider and can provide specific insight....we'r eshooting blind. |
| 21:37.55 | xai | Im trying to compile 1.2.17 on RH 4.2, 2.6.9-22EL kernel: I get this: chan_zap.c:9187: error: structure has no member named `call' |
| 21:38.10 | Toli_ | I use as provider 2 VoipCheap and VoipTrading |
| 21:38.13 | Mercestes | xai: Did you compile zaptel first? |
| 21:38.24 | elriah | For some reason, my voicemail recordings aren't playing. The callers are getting the message beep. the files are there and record properly. Any suggestions? |
| 21:38.29 | Mercestes | Toli_, never heard of 'em. |
| 21:38.34 | Toli_ | lol oj |
| 21:38.36 | Toli_ | ok |
| 21:38.49 | Mercestes | elriah, play the file manually as root and see if they have sound. |
| 21:39.14 | Mercestes | elriah, Given that case they may be being recorded in a codec that * cannot playback, or it could be a read permissinos issue. |
| 21:39.22 | elriah | Mercestes: The temporary message does. |
| 21:39.23 | elriah | hrm. |
| 21:39.27 | xai | Mercestes: no I did not.. |
| 21:39.46 | Mercestes | xai: Do you intend to compile zaptel? |
| 21:40.30 | xai | Mercestes: im not sure, we are all IP, but I'll compile it if I need to.. Didn't see that in the Readme's or help files. |
| 21:41.20 | mcab | thekidrio: which polycom phone is this? |
| 21:41.27 | xai | Mercestes: is that required for non zaptel equipment? |
| 21:41.48 | thekidrio | mcab, no model name yet, just announced on M$ newsfeed |
| 21:41.54 | thekidrio | lemme see if I can grab a link/pic for you |
| 21:42.18 | Mercestes | xai: I don't know why you have chan_zap.so failing to compile into asterisk when you are not compiling zaptel... |
| 21:42.34 | mcab | thekidrio: thanks |
| 21:42.40 | Mercestes | xai: howevre, I suggest you *do* compile zaptel with "ztdummy" uncommented so meetme, conferencing, and queues will work. |
| 21:43.03 | Mercestes | xai: And that should correct your chan_zap compile failure. You probably have a flag/switch/setting wrong in your MAKEFILE or yo ufailed to run ./configure or something. |
| 21:43.38 | xai | Mercestes: there is no configure file in the tarball.. |
| 21:44.08 | Mercestes | xai: Are you following a howto? |
| 21:45.04 | thekidrio | http://blog.tmcnet.com/blog/tom-keating/microsoft/office-communications-server-2007-public-beta-launches.asp |
| 21:45.15 | thekidrio | mcab, there ya go some neat features |
| 21:45.16 | xai | Mercestes: no I am not.. |
| 21:45.21 | Qwell[] | yay, now we can *hear* a blue screen |
| 21:45.31 | thekidrio | hahahaha |
| 21:45.31 | mcab | thekidrio: thanks, having a look |
| 21:45.39 | thekidrio | what is the sound of a BSOD? |
| 21:45.45 | thekidrio | bill gates laughter? |
| 21:45.51 | thekidrio | or woz weeping? |
| 21:45.55 | xai | Mercestes: just the info files.. |
| 21:47.44 | Mercestes | xai: I would suggest a google of asterisk install. Go ahead and DL zaptel and compile it in first. |
| 21:48.07 | mcab | iiiinteresting... |
| 21:49.44 | thekidrio | neat looking phones eh |
| 21:49.48 | fr0z3n | hey guys...can anyone look at this and help me figure out wat the problem is: http://pastebin.ca/424577 |
| 21:50.40 | xai | Mercestes: should I get the 1.2.16 version for ast 1.2.17 ? |
| 21:52.12 | *** join/#asterisk Innatech (n=daf@netblock-72-25-97-119.dslextreme.com) |
| 21:52.45 | *** join/#asterisk ValDuane (n=valduane@cpc4-basf6-0-0-cust470.nott.cable.ntl.com) |
| 21:53.01 | javb | Hi guys... i have 2 lines, form different service providers.. when trying to take a call out from one, 60% of the time, it gives me the error of area code.. like it is taking appart the first digit.. |
| 21:53.04 | javb | im seeing in asterisk console it is dailing correct... the big problem is that the error is not always.. it is as i said, 60 % of the time. |
| 21:53.18 | javb | othewise, using the other service provider, the is NO PROBLEM at all. |
| 21:53.19 | ber_ | is there any way to play a wav/mp3 from an offset with asterisk |
| 21:53.22 | ValDuane | anyone here who can help me install my TE205P on my trixbox? Having endless problems with the driver |
| 21:53.22 | javb | any ideaS? |
| 21:53.24 | ber_ | (skip 1 min into file) |
| 21:53.53 | javb | (TDM400P) |
| 21:53.59 | ber_ | javb you can try prepending a digit to that carrier |
| 21:54.06 | ber_ | prepend one |
| 21:54.13 | *** join/#asterisk Exc3ss_ (i=sdhdghs@host210-115-dynamic.57-82-r.retail.telecomitalia.it) |
| 21:54.27 | javb | ber_ , what do u mean with prepending? |
| 21:54.36 | ber_ | what number are you sending to that carrier |
| 21:54.52 | ber_ | like 4157773456? |
| 21:54.52 | *** part/#asterisk JoeDeveloper (n=jdevel@www.airlinksystems.com) |
| 21:54.53 | javb | like 809 569 3995 |
| 21:54.58 | ber_ | or 14157773456 |
| 21:55.02 | javb | both |
| 21:55.14 | ber_ | you should standardize what you are sending to one format |
| 21:55.15 | javb | depends.. if it is a cell phone or a residential phone |
| 21:55.41 | ber_ | most carriers I have dealt with (wireline) expect a certain number of digits |
| 21:55.46 | xai | I suppose the minor number of zaptel matches the minor number of *.. |
| 21:55.48 | ber_ | and will cause issues if you send them too many or too few |
| 21:56.07 | javb | so... details please.. kind of new. |
| 21:56.12 | fr0z3n | can anyone look at this error and help me figure out the problem pls... http://pastebin.ca/424577 |
| 21:56.15 | *** join/#asterisk mxmasster (n=Max@129.47.12.101) |
| 21:56.17 | mxmasster | hi all |
| 21:56.52 | mxmasster | i need some help configuring a cisco 5350 gateway to asterisk, anyone here using these things? |
| 21:56.58 | ber_ | i dont know exactly how to do it in asterisk |
| 21:57.16 | ber_ | but there might be macros on voip-info.org to do it |
| 21:57.23 | ber_ | i.e. normalizing dialed numbers |
| 21:57.42 | ber_ | you would have some code in your outbound dialplan to normalize the digits you are sending to your carrier |
| 21:57.50 | ber_ | its just simple text processing for asterisk |
| 21:58.10 | javb | ber_, but cant understand what means normalize yet :S |
| 21:58.32 | *** join/#asterisk |dennis| (n=dennis@shc.edu.bz) |
| 21:59.01 | ber_ | normalize means if you have someone dial 9777FILM 914157773456 and 94157773456 you send the carrier 4157773456 in all instances |
| 21:59.40 | ber_ | another way to accomplish this is to make the match on your outbound calls so it only makes the calls if they match the NPANXXXXXX format |
| 21:59.53 | ber_ | so if someone dials 14157773456 it would reject |
| 21:59.56 | ber_ | as well as 7773456 |
| 22:00.01 | javb | thats what i am doing ber_ ... people here just dial the SAME WAY the carrier will accept it.. if u take an analog normal phone, u dial 8095693995 ... same with asterisk. |
| 22:00.02 | ber_ | and only permit dialing of 10 digit |
| 22:00.57 | xai | Is the tarball for 1.2.17 supposed to come with a configure script? |
| 22:01.01 | ber_ | are these sip carriers or wireline carriers |
| 22:01.01 | javb | if u call a cellphone, y dial 18091234567 |
| 22:01.15 | javb | wireline carriers. |
| 22:01.22 | javb | TDM / SS7 |
| 22:01.28 | javb | im using TDM400P |
| 22:02.34 | javb | if u dial, it will go OK 50 % of the calls. |
| 22:02.34 | javb | 10 times u dial. 5 like eats the first digit |
| 22:02.34 | ber_ | asterisk eats the first digit or the carrier does? |
| 22:02.48 | javb | but, with another provider i have here, using the same dialplan, jjust changing cables.. wont happen. |
| 22:02.53 | Toli_ | Mercestes We tried something else, all the Sip.conf are made like "allow=g729" when I change it to "allow=all" we have the same error as from the 2nd provider but with the first |
| 22:03.24 | javb | ber_ y say Asterisk because the other carrier wont do it, and using a normal analog phone that "!@#$" carrier will let the call Ok |
| 22:03.39 | *** join/#asterisk Cresl1n (i=matt@nat/digium/x-3a50938259c213c1) |
| 22:03.39 | *** mode/#asterisk [+o Cresl1n] by ChanServ |
| 22:03.52 | javb | note: imcoming calls are handled perfect. |
| 22:03.54 | ber_ | so if you plug the analog line into a handset |
| 22:04.00 | ber_ | for the carirer with 50% call completion |
| 22:04.06 | ber_ | and dial the same numbers exactly how asterisk does |
| 22:04.12 | ber_ | do the calls complete 100%? |
| 22:04.18 | javb | YES. |
| 22:05.23 | ber_ | does asterisk logging say anything about why those calls are dropped? and what do the CDR say |
| 22:05.50 | cv_guest | is anyone famliar with unlocking a motorola vt1000 from vonage |
| 22:05.51 | xai | In function `pri_dchannel': chan_zap.c:9187: error: structure has no member named `call' |
| 22:06.29 | xai | Mercestes: i installed zap 1.2.16, but asterisk still gives me similar errors on make. |
| 22:07.10 | wunderkin | javb, using analog, you may need to add a small wait before it dials... zap/1/www1234567890 |
| 22:07.22 | javb | the call is NOT being dropped. Asterisk is dailing the call, but the service provider is saying.. in the case of dailing a cellphone (with 1 at the beggining) "please, for cellphone network u need '1' plus the number"... again, this is not ALL the time.. otherwise.. if it is 8091234567 ... carrier says "please, remember to dial '809' for a local call' |
| 22:07.33 | javb | wunderkin... /www ? |
| 22:09.16 | Toli_ | ~allow |
| 22:09.28 | javb | wunderkin , i have exten => _XXXXXXXXXX,1,Dial(Zap/3/${EXTEN}) |
| 22:09.36 | fr0z3n | can anyone look at this error and help me figure out the problem pls... http://pastebin.ca/424577 |
| 22:09.43 | *** join/#asterisk yenno (i=yenno@84-72-188-127.dclient.hispeed.ch) |
| 22:09.58 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 22:10.16 | yenno | hi, which (do you think) is the best web interface for asterisk? |
| 22:10.38 | fr0z3n | == Starting SIP/1000-0879d360 at outbound,2000,1 failed so falling back to exten 's' |
| 22:11.22 | fr0z3n | i am trying to call Office Communicator (IM)...thats the error shows up in asterisk Console. I am able to call from phone to phone but not from the computer. |
| 22:11.50 | Toli_ | guys help me |
| 22:11.53 | *** part/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:12.10 | xai | maybe it is failing on something I don't need.. I see an asterisk executable. |
| 22:12.32 | wunderkin | javb, so add the www before ${EXTEN} |
| 22:12.44 | javb | just like that? |
| 22:12.48 | javb | 3 www ? |
| 22:13.42 | wunderkin | yes, you may have to add more or can use less |
| 22:14.02 | wunderkin | each w is a 0.5 sec wait i think |
| 22:14.58 | ValDuane | anyone here who can help me install my TE205P on my trixbox? Having endless problems with the driver |
| 22:14.58 | Mercestes | ~trixbox |
| 22:15.07 | jbot | i guess trixbox is junk - avoid. It is also unable to be supported here. Try joining #freepbx and asking there, or use the trixbox forums at http://www.trixbox.org/modules/newbb/, or known as 'sh1tbox', STAY AWAY! |
| 22:16.23 | Toli_ | Guys what is the problem with the codecs ??? I tried to change it and it works for 1 but not for other !!!! |
| 22:17.43 | *** part/#asterisk brodiem (n=brodiem@67.110.68.66.ptr.us.xo.net) |
| 22:18.17 | Toli_ | <Mercestes> Are you here ?? |
| 22:19.04 | irule | how can I make sure 0 goes to operator and 01XXXXXX makes an international call |
| 22:19.11 | _VoicemeUpDotCom | dont use trixbox |
| 22:19.15 | irule | propperly? |
| 22:19.18 | *** part/#asterisk mxmasster (n=Max@129.47.12.101) |
| 22:19.25 | _VoicemeUpDotCom | and if you have to then go to #trixbox |
| 22:20.01 | javb | wunderkin... THANKS A LOT |
| 22:20.06 | _VoicemeUpDotCom | i love jbot lol it just explained everything so brightly... |
| 22:20.09 | javb | ber_ thanks to you as well. |
| 22:20.15 | Mercestes | Toli_, I am no wwhat's up? |
| 22:20.48 | Toli_ | <Mercestes> I changed the codecs all comes from there I thing |
| 22:20.56 | Mercestes | Toli_, never allow=all. |
| 22:21.05 | Mercestes | Toli_, Bad.... |
| 22:21.20 | Mercestes | xai: google redhat asterisk install and find a howto. |
| 22:21.52 | Toli_ | <Mercestes> I did allow=all and with x-lite I can call from prov1 and prov2 with no problems !!! strange no ?? |
| 22:22.14 | *** join/#asterisk frenzy (n=frenzy@unaffiliated/frenzy) |
| 22:22.17 | Mercestes | Toli_, .....do not use allow=all. |
| 22:22.23 | frenzy | hi |
| 22:22.35 | Mercestes | Toli_, however, it does indicate there might be a codec problem...Do a show channels and see what codecs your working calls are using and allow taht codec |
| 22:22.35 | frenzy | are there any kind of known issues with1.2.17 |
| 22:22.46 | frenzy | whoops meaning with OH323 |
| 22:22.48 | Mercestes | frenzy, I heard a few mentions of seg-faults but nothing extravagent |
| 22:23.12 | frenzy | Asterisk and OH323 |
| 22:23.28 | Toli_ | <PROTECTED> |
| 22:23.28 | Toli_ | 0 active channels |
| 22:23.28 | Mercestes | frenzy, there *ARE* issues with OH323 don't use it. |
| 22:23.32 | frenzy | for some reason asterisk will stop and restart |
| 22:23.36 | Mercestes | Toli_, make a call first.... |
| 22:23.41 | Toli_ | ok |
| 22:23.45 | javb | is there a way to handle double lines in analog lines connected to my asterisk ... |
| 22:23.52 | javb | ? |
| 22:23.57 | frenzy | alternative for H323? |
| 22:23.59 | Mercestes | javb: Dial(zap/1/bob&zap/1/bill) |
| 22:24.02 | *** part/#asterisk xo8ox (n=pride_32@wsip-66-210-250-2.ph.ph.cox.net) |
| 22:24.03 | Mercestes | frenzy, SIP. |
| 22:24.07 | Mercestes | frenzy, Zap |
| 22:24.10 | Mercestes | frenzy, IAX2 |
| 22:24.15 | frenzy | I mean besides OH323 |
| 22:24.20 | frenzy | I need h323 |
| 22:24.27 | frenzy | my provider only supports tht |
| 22:24.27 | yenno | somebody knows if asterisk supports the cisco ip phones? |
| 22:24.58 | javb | Mercestes... can u explain |
| 22:25.10 | Toli_ | Channel (Context Extension Pri ) State Appl. Data |
| 22:25.10 | Toli_ | SIP/voicetrading-0c6a (default 1 ) Up Bridged Call SIP/113031-cd83 |
| 22:25.10 | Toli_ | SIP/113031-cd83 (spnet-out 003223774101 1 ) Up Dial SIP/003223774101@voicetrading|60 |
| 22:25.13 | Toli_ | sry |
| 22:25.35 | Mercestes | javb: Just & the two line stogteher to ring both. Dial(Zap/1/line1&Zap/1/Line2) |
| 22:26.01 | Toli_ | <Mercestes> thats what I see |
| 22:26.01 | Mercestes | Javb: & being quite literally "and" |
| 22:26.10 | Toli_ | <Mercestes> Cant see the codec |
| 22:26.16 | Mercestes | Toli_, Under show channels or sip show channels or show codecs or codecs or something along those lines it will list what codec a call is using. . Look for that. Asterisk CLI has a <tab> interface to help you see it. |
| 22:26.35 | Mercestes | frenzy, wglwat |
| 22:26.47 | Mercestes | frenzy, if it works now I wouldn't touch it anymore.....ever. just leave it alone. |
| 22:26.55 | javb | Damn, Mercestes, cant understand u pretty well... any how to or info in the web? |
| 22:26.55 | Toli_ | 10x |
| 22:27.10 | *** join/#asterisk flenders (n=fserto@unaffiliated/flenders) |
| 22:27.11 | Mercestes | javb: Are you trying to ring two lines??? |
| 22:27.44 | frenzy | I upgraded * to the latest 1.2 edition |
| 22:27.47 | frenzy | now it wont work |
| 22:27.54 | Mercestes | frenzy, Told you not to touch it |
| 22:28.12 | frenzy | well that was too late |
| 22:28.16 | frenzy | I did this couple days back |
| 22:28.17 | javb | Mercestes.. no.. An analog line what double line capacity... |
| 22:28.18 | Mercestes | I agree |
| 22:28.32 | Mercestes | javb: oh, so you want to HUNT across two lines outbound?? |
| 22:28.39 | javb | yes... |
| 22:29.02 | javb | in analog phones.. using flash.. u can change between lines. |
| 22:29.08 | javb | but how now?: |
| 22:29.11 | Mercestes | javb Exten => _X.,1,Dial(Zap/1/Bob) exten => _X.,2,Dial(zap/2/Bob) |
| 22:29.12 | javb | (new using Voip- |
| 22:29.23 | Mercestes | javb: Or you could set them in a group and dial the group. |
| 22:29.26 | xai | Mercestes: i found out i needed libpri and zaptel first before compiling asterisk. It compiled easily now. |
| 22:29.38 | Mercestes | xai: Awesome. Forgot about lib pri. |
| 22:29.43 | Mercestes | xai: see, even I can't do it from memory. ;) |
| 22:30.05 | javb | ...? |
| 22:30.17 | fr0z3n | can anyone look at this error and help me figure out the problem pls... http://pastebin.ca/424577 |
| 22:30.20 | fr0z3n | == Starting SIP/1000-0879d360 at outbound,2000,1 failed so falling back to exten 's' |
| 22:30.21 | Mercestes | Javb: follow the link |
| 22:30.22 | Mercestes | ~wiki |
| 22:30.26 | Mercestes | ~book |
| 22:30.33 | jbot | hmm... book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 22:30.33 | *** join/#asterisk ruied (n=ruied@bl7-218-157.dsl.telepac.pt) |
| 22:30.38 | Mercestes | ~docs |
| 22:30.40 | jbot | docs is probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com |
| 22:30.40 | javb | which link ? |
| 22:30.46 | Mercestes | Javb: the jbot links |
| 22:31.49 | Toli_ | <Mercestes> I did show codec the number 113031 and it shows me alot of codecs !! |
| 22:31.52 | *** join/#asterisk ObsidianX (i=evilmonk@psycho.pengu.in) |
| 22:32.22 | Toli_ | and when I did show codecs it show all the codecs allowed from asterisk |
| 22:32.47 | ruied | how do I know what signalling for my fxo (PSTN) I should use for my country? I have a digium tdm400, living in Portugal, but don't know where to find the proper signalling metlhod from my provider ("Portugal Telecom" provider) |
| 22:32.52 | *** join/#asterisk arrenlex (n=em@S01060040052da362.ed.shawcable.net) |
| 22:33.33 | arrenlex | I have a modem card. I want to plug a phone line into it and use my computer's headset to make calls over it, instead of a regular phone. Is this possible? Is asterisk the right tool for this? |
| 22:34.27 | Mercestes | toli |
| 22:34.34 | Toli_ | jaa |
| 22:34.40 | *** join/#asterisk CrazyTux (n=CrazyTux@64.95.219.140) |
| 22:34.43 | Mercestes | sip show channels. look for "Form" |
| 22:34.45 | bulle | arrenlex: i have no clue on what is physicaly possible to do with a modem, but, asterisk is not what you are looking for if all you want is to place some calls |
| 22:35.07 | ObsidianX | hey folks, are the txgain and rxgain options new to 1.4.x? |
| 22:35.34 | arrenlex | bulle: I see. Thank you. |
| 22:35.53 | *** join/#asterisk mog (n=mog@c-71-207-200-130.hsd1.al.comcast.net) |
| 22:35.53 | *** mode/#asterisk [+o mog] by ChanServ |
| 22:36.08 | Toli_ | <Mercestes> is it the format I should look at ??? |
| 22:36.14 | fr0z3n | anyone know what this means? : == Starting SIP/1000-0879d360 at outbound,2000,1 failed so falling back to exten 's' |
| 22:36.18 | *** part/#asterisk arrenlex (n=em@S01060040052da362.ed.shawcable.net) |
| 22:36.23 | Toli_ | its is unknow(d) |
| 22:36.36 | Mercestes | Toli_, Yes |
| 22:37.03 | Toli_ | <Mercestes> but it is unknow(d) what should it be ? |
| 22:37.16 | Mercestes | Toli_, try disallow=all allow=ulaw |
| 22:37.16 | Toli_ | <Mercestes> I did allow=GSM now |
| 22:37.22 | Toli_ | ok |
| 22:37.38 | flenders | hey, do you guys use DHCP or you set static IPs to the phones? |
| 22:38.01 | Mercestes | flenders, yes. |
| 22:38.12 | flenders | yes? which one? |
| 22:38.15 | flenders | dhcp? |
| 22:38.29 | Mercestes | flenders, yes to your question. We either use DHCP or set static IPs to the phones. |
| 22:38.48 | flenders | well, that's pretty obviuos |
| 22:39.10 | Mercestes | flenders, ok. Then you should be able to proceed accordingly, no? |
| 22:39.22 | Mercestes | flenders, Do you prefer to use DHCP or static addresses on your phones? |
| 22:39.46 | flenders | cause I had set static ips to all our phones here, but now, I need to swap a couple of handsets around, and I wanted to keep the IPs assigned to a person's phone, not the hardware itself |
| 22:40.03 | flenders | just wondering what people usually do |
| 22:40.08 | blackk | fr0z3n: couldn't find any priority 1 for any extension that matched with 2000 in outbound |
| 22:40.24 | *** join/#asterisk syzygyBSD (n=chatzill@66.226.228.204.cpe.speedyquick.net) |
| 22:40.42 | flenders | I'm thinking of adding all the MAC addresses to dhcpd.conf, and in there I can just swap MACs and restart the phones |
| 22:40.50 | Toli_ | <Mercestes> is the ulaw popular for most SIP Phones ??? |
| 22:40.54 | flenders | just wanted to make sure it was a smart move |
| 22:41.21 | fr0z3n | blackk: outbound is a context? i dont even know where its getting that from...i dont have a context named "Outbound" |
| 22:41.28 | Mercestes | flenders, You cannot swap macs on the phones. I would just reassign the static IPs. If you assign a static IP to a MAC address via DHCP it's still DHCP just statically assigned. |
| 22:41.59 | Mercestes | flenders, There are unique problems to both methods but both methods are valid and have their merits. |
| 22:41.59 | *** join/#asterisk Defraz (n=t0tal@fw.fuzecore.com) |
| 22:42.09 | Defraz | Is anyone using the polycom 501 phone? |
| 22:42.14 | syzygyBSD | where would you suggest I purchase a cisco 7960IP? |
| 22:42.19 | syzygyBSD | Defraz: yes |
| 22:42.22 | Mercestes | flenders, for large distributions tho, I tend to use DHCP with Option(150) enabled. Asterisk will tell me what IP is assigned to each phone quite effectively |
| 22:42.24 | flenders | Mercestes: yeah, but the same person would have the same IP, and I guess it would be easier to provision, no? |
| 22:42.25 | blackk | fr0z3n: you have the context set to 'outbound' for your extension 1000 or that's the default context defined |
| 22:42.26 | Defraz | I am having an issue with volume. I set the volume on th ephone per call and it resets back to what it was. |
| 22:42.29 | Mercestes | But I'v edone stati caddressing as well |
| 22:43.03 | *** join/#asterisk CrashHD (n=crashhd@c-67-166-155-233.hsd1.ca.comcast.net) |
| 22:43.16 | flenders | Mercestes: you know a user's phone address, and no matter what hardware this person has on his desk, the IP would be the same |
| 22:43.22 | fr0z3n | blackk: i am able to call from 1000 to 2000 and 2000 to 1000...but this error shows up when i am trying to call from LCS (Office Communication, IM messenger) to one of the extension |
| 22:44.06 | fr0z3n | can someone look at my sip.conf or extensions.conf to check what the problem could be? i just cant figure this out |
| 22:44.23 | blackk | fr0z3n: put it on a pastebin |
| 22:44.27 | fr0z3n | thnx |
| 22:44.28 | fr0z3n | 1 moment |
| 22:44.37 | Toli_ | <Mercestes> What if 1 Phone is set to g729 and other ulaw is there a problem if they call internaly ?? |
| 22:45.13 | Mercestes | Toli_, Yes, you need licenses to transcode |
| 22:45.18 | *** join/#asterisk joobie (n=joobie@joobie.org) |
| 22:45.22 | wunderkin | Defraz, admin manual.... |
| 22:45.25 | Mercestes | flenders, You'llh ave to change the static IP then |
| 22:45.28 | wunderkin | look for persist |
| 22:45.48 | syzygyBSD | can anyone recommend a good phone dealer? |
| 22:45.49 | bulle | Mercestes: license need depends on where you live though, right ? |
| 22:45.56 | Toli_ | <Mercestes> What should I do than ??? |
| 22:46.13 | Mercestes | bulle, no. It depends on if you steal or not. |
| 22:46.21 | bulle | Mercestes: huh ? |
| 22:46.28 | irule | how can I call myself in kphone/linux and not get alsa error? |
| 22:46.53 | bulle | Mercestes: if he lives in a country where you can not patent software, algorithms etc, he isnt doing anything bad |
| 22:47.29 | Mercestes | bulle: Since it was made in a country where you can patent software, it would be theft to use variants of that software that voilate that patent |
| 22:47.49 | Mercestes | bulle: So i fyou want support in this channel yo uhave to pay for yoru g729 licenses regardless of where you live. |
| 22:47.55 | Mercestes | bulle: so the answer to yoru question is "no." |
| 22:48.24 | Toli_ | So I should pay to use the codec, right ? |
| 22:48.25 | bulle | Mercestes: eh ? it doesnt matter if software algorithm A is patented in usa or japan, as their law doesnt apply to belgium or wherever |
| 22:48.37 | joobie | Hey Guys.. I'm buying a server for my VOIP setup.. just wondering what I should make sure it has? |
| 22:48.38 | Mercestes | bulle: I'll remembre not to support you then. |
| 22:48.43 | joobie | I was going to go for a standard P4 3GHz |
| 22:48.46 | joobie | 1GB of RAM |
| 22:48.48 | Mercestes | bulle: i realyl don't care what your ethics are, nor am I the morality police... |
| 22:48.55 | joobie | and 80GB HDD |
| 22:49.07 | Mercestes | nor do I want to flesh out an international war here in #asterisk... |
| 22:49.10 | bulle | Mercestes: you are, just decided that you have the moral right to tell me what i can and can not do |
| 22:49.18 | Mercestes | but i fyo uwant g729 support in #asterisk you have to pay for the licenses. |
| 22:49.24 | bulle | Mercestes: or what Toli_ can and can not do |
| 22:49.26 | Mercestes | you can do whatever the hell you want. |
| 22:50.05 | Mercestes | I can't *stop* you from installing non-legit copies of G729...but I can ignore you if you do. |
| 22:50.09 | Toli_ | Mercestes: But if I use 1 of dose codecs I have to pay right ? |
| 22:50.20 | bulle | Mercestes: except that the code isnt non-legit for Toli_ |
| 22:50.29 | Mercestes | Toli_, you only have to pay to transcode. g729 to g729, iirc is free. |
| 22:50.39 | Mercestes | begone, degenerate. |
| 22:50.44 | joobie | heh |
| 22:50.57 | Mercestes | Toli_, to transcode from ulaw to g729 or vice verse there is a licensing fee. |
| 22:51.00 | thekidrio | joobie, fonality has some good info on the hardware to handle calls |
| 22:51.13 | thekidrio | lemme find it one sec |
| 22:51.15 | Mercestes | Toli_, two, in fact, to go from g729 to ulaw, and back from ulaw to g729 |
| 22:51.27 | Toli_ | Mercestes: I see |
| 22:51.33 | Corydon-w | One per leg |
| 22:51.41 | bulle | Mercestes: im not trolling, im just trying to point out that the law is far from uniform, its basicly usa and japan that have a patent system that allows patents of software, most of the rest of the world doesnt, so claiming its illegal to someone not living in those countries, is just plain wrong |
| 22:51.46 | Toli_ | Mercestes: and I will have to install that transcode ?? right ? |
| 22:51.57 | *** join/#asterisk wunderkin- (n=kev@dslstat-ppp-95.fastq.com) |
| 22:52.07 | Mercestes | bulle: I don't give a damn what your law says or where you are from..... |
| 22:52.10 | thekidrio | http://www.fonality.com/pop.cgi?page=which_server_is_right_for_me.tt |
| 22:52.10 | Corydon-w | bulle: it's also illegal for us to help you infringe a patent |
| 22:52.15 | joobie | thekidrio, thanks :) |
| 22:52.24 | thekidrio | np it helped me hehe |
| 22:52.28 | joobie | wow a .tt page |
| 22:52.30 | joobie | wat's that? |
| 22:52.33 | joobie | +h |
| 22:52.38 | syzygyBSD | can anyone explain to me the difference between a cisco 7960IP and 7960G? |
| 22:52.41 | thekidrio | you know I am not sure haha |
| 22:52.41 | Mercestes | Toli_, yes, or use all ulaw or g729 |
| 22:52.45 | bulle | Corydon-w: intresting, and what country do you live in then, usa ? |
| 22:52.52 | joobie | heeh |
| 22:52.54 | bulle | Corydon-w: and no, i dont use g729 myself |
| 22:52.56 | joobie | yea first time i've seen it too :P |
| 22:52.58 | Corydon-w | bulle: correct |
| 22:53.20 | Corydon-w | bulle: it's called contributory infringement, and it gets pretty expensive |
| 22:53.27 | joobie | hey thekidrio, what does it mean by "ACD Calls" ? |
| 22:53.30 | bulle | Corydon-w: ok, then i understand why Mercestes is not willing to help, but i think it would be better to explain the situation instead of just saing its illegal |
| 22:53.43 | blackk | bulle: just curios, can you give us some examples of coutries where you cannot patent software/algorithms? |
| 22:53.56 | Mercestes | Belgium obviously. |
| 22:54.01 | JunK-Y | automatic call distribution |
| 22:54.01 | bulle | blackk: you cant enforce software patents in the eu |
| 22:54.08 | joobie | what's that? |
| 22:54.17 | bulle | blackk: same goes for nearly all of south american and africa |
| 22:54.19 | Corydon-w | Actually, Belgium is under the EU, and the EU is okay with software patents |
| 22:54.23 | Toli_ | can you tell me where to buy that license ? |
| 22:54.30 | voipman | anyone have a recommendation on the maximum ulimit -c value in asterisk startup script? 8096 doesnt seem enough for > 1000 calls |
| 22:54.33 | *** join/#asterisk dimmik (n=dimmik@static217244.dsl.hol.gr) |
| 22:54.36 | bulle | Corydon-w: no, you cant have pure software patents in eu, they are unenforcable |
| 22:55.10 | bulle | Corydon-w: the idea was that soft patents would be allowed, but the proposal was voted down and is no on some sort of redesign |
| 22:55.16 | bulle | s/no/now/ |
| 22:55.31 | Corydon-w | bulle: I am no lawyer, nor do I play one on TV. At the point where a lawyer says "you need to do this to be legal", that's what I do. |
| 22:55.47 | thekidrio | haha the bot switches for yas eh |
| 22:55.56 | bulle | Corydon-w: yes, and since you explained why, i fully understand you |
| 22:55.57 | Mercestes | I realy don't see how that makes i tethical to say, take code from patented countires and redo it in a "no patent" country and distribute it without patents and call that legal. |
| 22:56.01 | thekidrio | Corydon-w, never trust a lawyer |
| 22:56.08 | *** join/#asterisk haroldp (n=nnnnnnnn@taurus.coyotecraft.com) |
| 22:56.20 | bulle | Mercestes: because us law doesnt have power over anything but us ? |
| 22:56.24 | Corydon-w | thekidrio: I trust lawyers that I pay to represent my interests |
| 22:56.25 | haroldp | hullo |
| 22:56.29 | thekidrio | Mercestes, during the cold war america did not respect soviet copyrights and vice versa |
| 22:56.33 | Mercestes | bulle: So it's ok to steal from the US and ignore our laws and customs? |
| 22:56.42 | thekidrio | Corydon-w, those i trust the least heh |
| 22:56.50 | Corydon-w | Mercestes: quit with the trolling |
| 22:56.56 | Mercestes | ...how am I trolling? |
| 22:56.59 | dimmik | Is anyone familiar with astibank? I was just wondering does it do hw echo cancellation? |
| 22:57.02 | bulle | Mercestes: well, its perfectly legaly for you to behave barbaricaly and totaly ignore our laws and customs |
| 22:57.07 | thekidrio | so TV shows like the simpsons would use russian music and they would use our music neat eh |
| 22:57.30 | Mercestes | Why don' tyou talk to thse other jokers? I'm trying to answer asterisk question sover here and bulle is butting in with his crap. |
| 22:57.31 | bulle | thekidrio: pretty funny actualy, i didnt know that |
| 22:57.51 | thekidrio | yeah good way to save money |
| 22:57.51 | Corydon-w | bulle: I dunno, the point at which you believe the US would never kidnap a foreign national and stick him in Gitmo is the point at which you can violate patents freely |
| 22:58.20 | bulle | Corydon-w: huh, not sure i understood that |
| 22:58.33 | joobie | hmm.. how much HDD storage would I need |
| 22:58.41 | bulle | Corydon-w: but i guess its a bit off topic realy |
| 22:58.46 | joobie | I mean say I have 30 users.. for voicemail and asterisk, how much is the average? |
| 22:59.15 | joobie | I was just going to get an 80GB in a RAID1 configuration, but not sure if this will be enough? |
| 22:59.21 | dimmik | 1meg/min or so |
| 22:59.29 | Corydon-w | bulle: it's an extreme case, but the US has been known to be rather barbaric against people who violate their laws, even internationally. |
| 22:59.40 | Corydon-w | bulle: the question is, are you comfortable with the risk? |
| 23:00.01 | joobie | dimmik, so you're thinking more so the voicemail, how muhc it will chew up? |
| 23:00.05 | bulle | Corydon-w: yes |
| 23:00.20 | thekidrio | hehe |
| 23:00.30 | thekidrio | mmm risk assessment |
| 23:00.56 | bulle | Corydon-w: i seriously doubt cia or homeland security or whatever will come with black choppers and pick Toli_ up for not doing anything criminal, in his home country |
| 23:01.05 | dimmik | joobie: 80 gb is plenty |
| 23:01.09 | joobie | hey dimmik is RAID1 ok for asterisk? does it need high disk performance? |
| 23:01.14 | joobie | thanks dimmik |
| 23:01.26 | joobie | I just did some basic math of your estimate.. 1GB = 16hrs of playback |
| 23:01.28 | thekidrio | i use raid 5+1 |
| 23:01.40 | *** join/#asterisk bawb2 (n=bawb2@ip51051.estcmp.ku.edu) |
| 23:01.40 | joobie | I just wanted to keep costs down |
| 23:01.42 | joobie | so RAID1 would be cheaper.... |
| 23:01.55 | joobie | but if i skimp on it and get the RAID1, duno if it'll be sufficient |
| 23:02.03 | bulle | joobie: new harddrives, can easily write 20 to 30 megabytes per second, a phonecall is, say 64kbit |
| 23:02.19 | thekidrio | and harddrives are getting cheaper and cheaper joobie |
| 23:02.23 | bulle | joobie: now it gets more complicated when there are lots of reads and writes at the same time |
| 23:02.24 | robin_sz | joobie, for info, as I remember, some test were doen with a very basic 500mhz box and asterisk |
| 23:02.25 | menfin | i've just connected my first analogic phone on my asterisk ipbx and i can listen to the demo music |
| 23:02.27 | thekidrio | you can always buy another |
| 23:02.43 | robin_sz | joobie, they tried with around 700 users if I remember right, and it was fine |
| 23:03.05 | thekidrio | 700? how many at the same time thats the question heh |
| 23:03.16 | bulle | thekidrio: indeed |
| 23:03.24 | joobie | see this is for a client |
| 23:03.35 | joobie | i have to give him the hardware saying it will last yoiu up to .. 50 users for example |
| 23:03.45 | joobie | i cant give it and if it doesn't last get him to upgrade again |
| 23:03.56 | thekidrio | I am setting up some 300 chair call centers i don't think that 500mhz box would work hehe |
| 23:04.01 | robin_sz | well is it more than 500mhz? |
| 23:04.04 | bulle | joobie: well, as thekidrio said, the question is, how many users at one time ? |
| 23:04.05 | joobie | so i can a) go straight for the RAID5 and potentially charge a bit more that is not needed or b) get RAID1 and cross my fingers? |
| 23:04.12 | joobie | well say 25 users at one time |
| 23:04.24 | thekidrio | joobie, tell him he has X amount of hours at Y amount of space |
| 23:04.38 | joobie | the hours / space will be fine though |
| 23:04.42 | joobie | i mean 16hrs for 1GB |
| 23:04.49 | thekidrio | yeah the rest of your box is fine |
| 23:04.50 | *** join/#asterisk [-Quasar-] (n=jelkj@h8441149158.dsl.speedlinq.nl) |
| 23:04.50 | joobie | 80GB in a RAID1 will be more than sufficient |
| 23:05.02 | thekidrio | unless its a constant call environ |
| 23:05.05 | CrashHD | in the cdr records (mysql) is there a way to have a unix timestamp rather than calldate? or in conjuction with? |
| 23:05.08 | dimmik | joobie: disk i/o should not be an issue |
| 23:05.14 | joobie | ok |
| 23:05.18 | joobie | so RAID1 is fine? |
| 23:05.24 | thekidrio | the last job I had did 18 on constant dial and 50 on roughly 10 dials an hour |
| 23:05.25 | dimmik | sure |
| 23:05.30 | joobie | also guys.. dimmik that link you gave me said get a XEON For 45ish calls |
| 23:05.35 | joobie | i was hoping to just get a p4 :/ |
| 23:05.40 | joobie | that sucks!:P hehe |
| 23:05.43 | thekidrio | 45 ACD calls right? |
| 23:06.21 | joobie | 25 ADC, 45 CALLS |
| 23:06.21 | dimmik | joobie: what link? |
| 23:06.21 | joobie | http://www.fonality.com/pop.cgi?page=which_server_is_right_for_me.tt |
| 23:06.21 | thekidrio | http://www.fonality.com/pop.cgi?page=which_server_is_right_for_me.tt |
| 23:06.22 | thekidrio | that link |
| 23:06.22 | thekidrio | heh |
| 23:06.22 | joobie | oh sorry |
| 23:06.23 | joobie | thekidrio sent it even |
| 23:06.23 | thekidrio | my bad, he asked you |
| 23:06.26 | joobie | sorry, too early in the morning |
| 23:06.32 | dimmik | :) |
| 23:06.35 | joobie | just rolled outa bed and jumped into #asterisk ;P |
| 23:06.38 | thekidrio | i am 12 mins away from leaving for the day hehe |
| 23:06.48 | joobie | okok on emore question |
| 23:06.58 | joobie | heehe thekidrio |
| 23:07.01 | robin_sz | pah, what rubbsh |
| 23:07.14 | joobie | what sort of "expansion slots" should i be looking for in the server I buy? |
| 23:07.18 | joobie | jusrt standard PCI slots? |
| 23:07.22 | thekidrio | pci 2.2 compliant |
| 23:07.24 | joobie | PCI-X ?? |
| 23:07.38 | robin_sz | how do you expect to connect to the phone network? |
| 23:07.39 | thekidrio | if you are going to use TDM or T1/PRI stuff |
| 23:07.50 | joobie | is it just for the phone network? |
| 23:07.51 | thekidrio | otherwise you don't need any expansion slots |
| 23:08.04 | thekidrio | yeah |
| 23:08.04 | dimmik | joobie: If you don't have transcoding, etc np with a p4. Go for more pci slots since most cards are pci |
| 23:08.04 | joobie | i mean the other end for my LAN that uses VOIP, that comes out of a standard ethernet port right? |
| 23:08.26 | thekidrio | umm it can yeah |
| 23:08.30 | joobie | hmm not sure what transcoding is |
| 23:08.31 | thekidrio | mine is fiber hehe |
| 23:08.35 | joobie | nice |
| 23:08.47 | thekidrio | transcoding is when say an mp3 needs to become an mp4 |
| 23:08.51 | joobie | ahhh |
| 23:08.51 | thekidrio | and vice versa |
| 23:08.53 | robin_sz | well, I have 40 users on a old 600mhz P3 and the load figures are so low, you can barely see them |
| 23:08.53 | thekidrio | at the same time |
| 23:09.02 | joobie | but cant i just write it out in its native format? |
| 23:09.07 | joobie | what's the purpose to transcode? |
| 23:09.08 | thekidrio | robin_sz, 40 users on constant calls? |
| 23:09.20 | thekidrio | if one phone uses one codec and the other uses another |
| 23:09.30 | thekidrio | sorry if one ENDPOINT |
| 23:09.34 | thekidrio | does not need to be a phone |
| 23:09.37 | robin_sz | obviously, we try not to transcode |
| 23:09.48 | robin_sz | like, duh! |
| 23:09.50 | thekidrio | robin_sz, hehe yeah |
| 23:09.58 | joobie | ahh |
| 23:09.58 | thekidrio | that cranks up top |
| 23:10.06 | robin_sz | indeed. |
| 23:10.06 | bulle | joobie: if one of your phones uses ulaw codec and some other uses say gsm, then transcoding is needed, to get the two to send data they both understand |
| 23:10.06 | thekidrio | but sometimes you have too |
| 23:10.11 | joobie | what ist he native format asterisk uses without transcoding? |
| 23:10.20 | robin_sz | any |
| 23:10.31 | thekidrio | it steps through the codecs i think |
| 23:10.33 | joobie | ahh i see |
| 23:10.42 | joobie | so asterisk writes to a particular format without transcoding |
| 23:10.50 | dimmik | It would be nice if * had a more clever codec negotiation algo |
| 23:10.53 | joobie | and you only need it if you then have inconsistancy throughout your network? |
| 23:11.02 | robin_sz | our outgoings are isdn a-law, so we use a-law everywhere .. seems best |
| 23:11.06 | thekidrio | dimmik, yeah did you read about microsofts codec? |
| 23:11.13 | joobie | that should not be a problem, im going to buy all the same phones |
| 23:11.23 | *** join/#asterisk ctaloi (n=ctaloi@pool-72-90-82-84.syrcny.fios.verizon.net) |
| 23:11.43 | dimmik | thekidrio: no, what's to read? |
| 23:12.03 | thekidrio | it sounds VERY clear and fairly small footprint |
| 23:12.05 | thekidrio | lemme find url |
| 23:12.37 | robin_sz | and the source code for this codec is available? |
| 23:12.50 | thekidrio | http://www.asteriskblog.com/microsoft-announces-office-communications-server-2007-public-beta/ and for pictures of the phone http://blog.tmcnet.com/blog/tom-keating/microsoft/office-communications-server-2007-public-beta-launches.asp |
| 23:12.51 | dimmik | but it's propriatery, right? |
| 23:13.01 | thekidrio | well yeah its microsoft heh |
| 23:13.09 | thekidrio | but it shows what a good codec can do |
| 23:13.15 | thekidrio | and that often pushes the open source community |
| 23:13.20 | robin_sz | g723 is good too |
| 23:13.37 | thekidrio | anyone here use speex on an enterprise level? |
| 23:13.37 | ctaloi | hey all, i had some thoughts about running two asterisk servers in tandem with a shared NFS to hold the configs, libs, and astdb - is this something that is even feasible ? |
| 23:14.08 | dimmik | I hear that the have excellent error consealment algo |
| 23:14.29 | thekidrio | you can supposedly use the M$ codec just fine over dial up |
| 23:14.39 | thekidrio | thats really what I am interested in |
| 23:14.52 | thekidrio | even if dial up is dying |
| 23:14.59 | joobie | guys just thinking about hardware.. i don't want to miss naything, can someone correct me if ive missed anything please.... 1) Server (to host asterisk) 2) VOIP Phones 3) PoE Switch 4) Cabling (is this just standard CAT5 when using PoE? 5) Interface Card to Telco |
| 23:15.03 | joobie | is that all that's needed? |
| 23:15.36 | dimmik | power plugs etc :)) |
| 23:15.40 | joobie | eheh |
| 23:15.43 | joobie | ahh UPS! |
| 23:15.44 | thekidrio | you don't NEED PoE, but it sure is nice |
| 23:15.45 | joobie | good point |
| 23:15.57 | thekidrio | and are you getting two of them? |
| 23:16.05 | thekidrio | one as the main server |
| 23:16.30 | thekidrio | then one in a geographically different place |
| 23:16.33 | dimmik | poe is expensive if u r trying to cut down cost |
| 23:16.35 | thekidrio | yeah |
| 23:16.39 | ctaloi | has anyone tried using a shared nfs w. 2 asterisk servers reading from it ? |
| 23:16.42 | joobie | i onyl wanted PoE so that I could make the UPS for the phones centralized |
| 23:16.48 | thekidrio | PoE is very costly for large networks |
| 23:16.52 | thekidrio | not too bad for new/small networks |
| 23:17.01 | thekidrio | refitting old networks sucks |
| 23:17.32 | dimmik | it's just a matter of whether have ups power plugs on every office |
| 23:18.23 | thekidrio | we got lucky with our new office, each station had 4 ethernet drops and 6 power outlets already on UPS in the server room |
| 23:18.33 | *** join/#asterisk Toli_ (n=toli@189.195-241-81.adsl-dyn.isp.belgacom.be) |
| 23:18.35 | thekidrio | i almost broke my spine replacing the batterys in the ups |
| 23:18.39 | thekidrio | err batteries |
| 23:18.44 | dimmik | If you still need poe, dlink makes a nice & cheap switch |
| 23:19.11 | thekidrio | anyone here use Nortel phones with a unistim server? |
| 23:19.22 | thekidrio | I just inherited several boxes of nortel phones |
| 23:19.30 | thekidrio | not sure if I want to try to just sell them off or get them working |
| 23:19.44 | thekidrio | but i have never used unistim heh |
| 23:21.17 | *** join/#asterisk ptblank (n=MURDER1@cpe-75-84-58-152.socal.res.rr.com) |
| 23:21.19 | thekidrio | ack I am supposed to be out of here. cyas everyone |
| 23:21.21 | joobie | sorry phone brb |
| 23:22.52 | ctaloi | humm - i've searched the list archives, and can't find much info on sharing diaplans etc.. over NFS, should I presume this is a bad approach to load balancing / redundancy ? |
| 23:23.00 | joobie | back |
| 23:23.18 | joobie | nope thekidrio, just one server.. why do I need two? |
| 23:23.35 | joobie | the problem with not going PoE is I have to patch in a UPS port to each desktop |
| 23:23.40 | joobie | otherwise in a power outage they all drop |
| 23:24.26 | joobie | why's PoE costly though? I thought the cost is just in the switch that support PoE which is around 700$ |
| 23:25.03 | dimmik | try dlink |
| 23:25.40 | haroldp | I installed asterisk. set up an extension in sip.conf. was able to connect with a sip phone and play the demo stuff. I set up a second extension, but I get a fast busy when I try to dial it. |
| 23:26.03 | haroldp | what do I need to do to make that work? |
| 23:26.08 | joobie | dimmik, did i forget anything though in that set, or that's pretty much all i need? |
| 23:26.20 | joobie | also does PoE work over the standard ethernet cable? |
| 23:27.08 | dimmik | yeap |
| 23:27.31 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 23:30.04 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-185-4.buckeyecom.net) |
| 23:31.21 | *** join/#asterisk ltdwk (n=z@203-173-10-9.perm.iinet.net.au) |
| 23:31.30 | *** join/#asterisk notoriousrab1982 (n=notoriou@207.47.34.74.static.nextweb.net) |
| 23:31.47 | dimmik | Has anyone used the octasic echocanceler? |
| 23:31.57 | notoriousrab1982 | i have a simple question about incoming calls, can anyone help me? |
| 23:32.09 | flenders | notoriousrab1982: post the question |
| 23:32.44 | notoriousrab1982 | flenders, thanks. i have set up a teliax account and have a DID, I can make calls go out, but dialling my DID, I cannot get the call to come into the asterisk box |
| 23:33.36 | notoriousrab1982 | i put the line exten => 9252712128,1,Answer() into my extensions.conf file under [incoming] context |
| 23:34.24 | notoriousrab1982 | if I look at teliax's web interface, you can see the call attempting to connect ie a cdr appears, but the call just disconnects |
| 23:36.09 | *** join/#asterisk ltd (n=z@202-161-20-161.dyn.iinet.net.au) |
| 23:37.53 | flenders | is teliax IAX? |
| 23:38.31 | notoriousrab1982 | its SIP - they provide sip origination and termination |
| 23:38.32 | notoriousrab1982 | they also do IAX, but I am only interested in using SIP at this time |
| 23:38.56 | flenders | paste your sip.conf on pastebin |
| 23:39.04 | flenders | ~pb |
| 23:39.50 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.ca, or http://channels.debian.net/paste |
| 23:40.30 | notoriousrab1982 | ok, will do |
| 23:41.10 | *** join/#asterisk mrpillowhead (n=pillowhe@ROAMFG-AUTHCLIENTS-01.NATPOOL.NYU.EDU) |
| 23:41.14 | notoriousrab1982 | http://pastebin.ca/424750 |
| 23:41.38 | mrpillowhead | what's a good IAX client for Fedora? |
| 23:42.32 | flenders | notoriousrab1982: is your asterisk answering the call and then dropping? or it's not answering at all? |
| 23:43.02 | notoriousrab1982 | i dont think it is answering at all, like I say, the CDR appears on the teliax website, but nothing appears on the asterisk console at all |
| 23:43.24 | flenders | oh, I see what you're missing |
| 23:43.41 | notoriousrab1982 | i have verbosity set high (well vvvvvv) and nothing appears, when I do internal calling, it usually produces a record |
| 23:43.45 | flenders | you said you created an extension 'exten => 9252712128,1,Answer()' on extensions.conf |
| 23:44.06 | flenders | your register line should have a '/9252712128' at the end... |
| 23:45.28 | notoriousrab1982 | ahh, will try that, thank you |
| 23:46.02 | bulle | notoriousrab1982: noticed that you got a private message ? |
| 23:46.48 | flenders | bulle: I did the same |
| 23:47.22 | flenders | people don't seem to remember that there are 300+ people here |
| 23:47.45 | bulle | flenders: ye, and there are always some bad seeds around |
| 23:48.25 | *** part/#asterisk fr0z3n (n=fr0z3n@CPE0016b64d8af3-CM000f9fa6b664.cpe.net.cable.rogers.com) |
| 23:48.32 | notoriousrab1982 | flenders: did not work, still the same, i do sip show registry and sip show peers and they are both registered ok to the teliax service, even though i added the 925 numberat the end |
| 23:49.03 | notoriousrab1982 | this could be a problem with teliax not routing that number to my asterisk box, what do you think |
| 23:50.14 | Mercestes | ping mog |
| 23:50.44 | bulle | notoriousrab1982: you can always turn on sip debuging and see what actualy happens when you try to place a call |
| 23:51.07 | haroldp | the error I get on console when i try to dial other extensions is: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 23:51.26 | notoriousrab1982 | thanks bulle - is that a console command? is it sip set debu |
| 23:51.28 | notoriousrab1982 | g |
| 23:52.23 | flenders | notoriousrab1982: you said you created the extension on context [incoming] |
| 23:52.34 | blackk | haroldp: you either don't have that extension define or the extension is unreachable (qualify=yes and it doesn't respond) |
| 23:52.42 | bulle | notoriousrab1982: yes, "sip set debug" |
| 23:52.45 | flenders | notoriousrab1982: but your [teliax] settings on sip.conf are 'context=xiptel' |
| 23:52.52 | flenders | they have to match |
| 23:53.35 | notoriousrab1982 | ahh, ok, I thought something did not make sense when it was put into incoming, because then it does not relate to the context that you are in |
| 23:53.36 | *** join/#asterisk Dr-Linux|home (n=asfdf@DSL-202-59-73-131.nexlinx.net.pk) |
| 23:53.43 | flenders | notoriousrab1982: do you have a context [xiptel] on extensions.conf? |
| 23:54.16 | notoriousrab1982 | flenders: yes, I do |
| 23:54.18 | flenders | notoriousrab1982: create that, or change the [teliax] context on sip.conf to incoming |
| 23:54.52 | flenders | make sure you do a 'sip reload' and 'extensions reload' on the CLI |
| 23:55.19 | flenders | notoriousrab1982: did you get our private messages? |
| 23:55.39 | bulle | notoriousrab1982: also, as i havent seen your extensions.conf, but, assuming that sip entry called 2615 is some local phone, then having both yoru local phone and your external sip provider going to the same context, might be a bit bad, as it might allow incomming calls to place outgoing calls |
| 23:55.52 | Dr-Linux|home | my asterisk server is behind the NAT, my asterisk sends calls through a sip provider, my problem is "one way audio" i can't listen other end |
| 23:55.56 | Dr-Linux|home | any suggestions? |
| 23:56.38 | bulle | Dr-Linux|home: 1) in sip.conf set the proper values for your public ip, eg externip |
| 23:56.39 | blackk | Dr-Linux|home: set nay=yes and externip and localnet in sip.conf |
| 23:56.56 | *** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net) |
| 23:57.11 | bulle | Dr-Linux|home: 2) if that doesnt help, then forward sip and rtp ports via router or whatever that has yoru public ip, to the asterisk box |
| 23:57.30 | bulle | Dr-Linux|home: you might want to narrow down the rtp portrange a bit, depending on how many concurrent calls you need etc |
| 23:57.59 | Dr-Linux|home | blackk: already tried all nat=options no luck |
| 23:58.16 | Dr-Linux|home | bulle: 1) every thing is proper |
| 23:58.49 | Dr-Linux|home | bulle: that makes sense, my router doesn't have capibility to forward the ports? :( |
| 23:58.58 | bulle | Dr-Linux|home: so, you have port 5060 and the rtp portrange forwarded to your asterisk box ? |
| 23:59.08 | bulle | Dr-Linux|home: no idea, but i bet it has |
| 23:59.36 | flenders | hey, who controls jbot? |
| 23:59.38 | bulle | Dr-Linux|home: just check the routers web interface, for "virtual server" "port forwarding" or similar named things |
