| 00:00.10 | *** part/#asterisk PanicMan (i=Learner@122.102.33.80) |
| 00:00.34 | layne | that's what happens when you wait 60 seconds |
| 00:00.53 | Deadpepsi | in this book it only says like 500 users?? wont this do 50,000 ext's |
| 00:01.35 | Winkie | layne: indeed |
| 00:01.52 | Winkie | Deadpepsi: asterisk will scale up appropriately with multiple servers, but it depends on how you set it up |
| 00:02.01 | Deadpepsi | ok |
| 00:03.21 | Deadpepsi | i was gonna use a like a amd quadcore with 16 gig of ram and a 1 TB hdd on a 20/20 firber liner |
| 00:04.05 | Winkie | Deadpepsi: what exactly are your plans? |
| 00:07.05 | Deadpepsi | for now a home system that will allow use to travle and use the sip phones like cells on open wifi 5 to 6 ext for now with 6 sip tunks |
| 00:07.45 | Deadpepsi | but i as i work and lean more i want to make so the rest of the family on other states and countries can use it as well |
| 00:08.05 | Deadpepsi | i dought i will have 50,000 exts and all |
| 00:12.45 | Deadpepsi | guess i went off the deep end already |
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| 00:21.26 | techie33 | does anyone one know how to change the defualt intercom answer timer from 5000 ms to something else? I have a cisco 7940 that will not answer in the allowed timeout period and just hangs up. |
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| 00:41.20 | dshap | i'm trying to set up my * box so that people can call into an extension, record an audio file, and then have the * box place a call to another phone and playback this audio file AFTER the initial caller has hung up...how would you guys do this? |
| 00:41.29 | dshap | would the best way be to call an AGI script that creates a .call file? |
| 00:45.26 | carrar | I could show you how I would do that |
| 00:45.35 | carrar | but then you would receive a bill |
| 00:45.42 | dshap | damn |
| 00:45.45 | dshap | i'm poor |
| 00:45.48 | dshap | this is a low-budget startup |
| 00:45.50 | dshap | any pointers? |
| 00:45.53 | s14ck | XD |
| 00:46.06 | carrar | use record |
| 00:46.09 | carrar | playback |
| 00:46.10 | dshap | carrar: is there a much better way than what i proposed? |
| 00:46.42 | carrar | there is always a better way |
| 00:47.26 | dshap | is there an asterisk concept/application that i'm likely not familiar with that i'd need for this task? |
| 00:47.32 | carrar | Whats is this for? paging? |
| 00:47.40 | dshap | no |
| 00:47.48 | dshap | it's for leaving people voicemails |
| 00:48.09 | dshap | you call in, record a message, and then the system will deliver the mssage as a voicemail to a different voicemail system |
| 00:48.33 | dshap | eventually i need to make it so that if a ton of poeple call in at once, the outgoing calls can be queued so that the number of concurrent outgoing calls doesn't bog the server down |
| 00:48.43 | dshap | which i know i'll have trouble with doing the call file way |
| 00:48.47 | dshap | since those all go out immediately |
| 00:49.36 | dshap | what should i read up on to learn how to do this? |
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| 01:23.34 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 01:23.34 | dshap | anything at all? |
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| 02:00.22 | cvnet | anyone here willing to help me setup a ipsec? I'm willing to pay your help |
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| 02:11.51 | haryv | seems my sendmail voicemail attacment went though but have not recived it yet in my yahoo email account. Usally a delay passing email though a smtp mail relay? |
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| 02:38.29 | toddejohnson | Can anyone tell me what "file.c: Unexpected control subclass '-1'" means? |
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| 03:03.59 | BeeBuu | how can i play a song from internet? |
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| 03:04.32 | Qwell | what? |
| 03:05.38 | [TK]D-Fender | BeeBuu: www.youtube.com |
| 03:05.51 | Qwell | res_youtubeonhold |
| 03:05.57 | Qwell | [TK]D-Fender: get on that |
| 03:06.39 | BeeBuu | [TK]D-Fender: i mean playback a song from internet |
| 03:06.41 | BeeBuu | in asterisk |
| 03:12.19 | MaliutaLap | Qwell: that's just sick, and twisted :P |
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| 03:15.41 | [TK]D-Fender | MaliutaLap: And you're just jealous he came up with it first :p |
| 03:16.40 | jaytee | I think res_lastfmonhold would be cooler |
| 03:17.16 | MaliutaLap | [TK]D-Fender: shush! I'll come up with plenty [more||of] sick and twisted things :P |
| 03:17.32 | MaliutaLap | how about res_devrandomonhold ? |
| 03:17.47 | MaliutaLap | although technically that could be done another way right now |
| 03:17.52 | jaytee | bet some companies have and S&T department along with and R&D department |
| 03:18.06 | *** join/#asterisk JackEStorm (n=no@ip24-252-118-155.no.no.cox.net) |
| 03:18.10 | MaliutaLap | the old cat /vmlinz > /dev/dsp |
| 03:19.34 | MaliutaLap | oh, and the option is back in kernel config to remove the protection that stops stupid users doing 'cat /dev/[random||zero] >/dev/mem' |
| 03:19.55 | MaliutaLap | considers release a branch of "sucker" kernel compiles |
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| 03:46.10 | *** join/#asterisk panicman (i=panicman@210.89.82.35) |
| 03:46.59 | panicman | i can change the outgoing caller ID,. but i need to change it ramdom basis, for each call, it will select the caller ID from a list |
| 03:47.05 | panicman | is it possible ? |
| 03:47.24 | panicman | any helper ? |
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| 03:48.18 | [TK]D-Fender | PanicMan: Its your dialplan do whatever you want |
| 03:48.32 | panicman | [TK]D-Fender >> Hi |
| 03:49.12 | panicman | selecting the callerID from a file is possible, right ? |
| 03:49.31 | panicman | like, 01211111111 to 01211111200 |
| 03:49.57 | panicman | each call will get the CID from that file |
| 03:50.06 | [TK]D-Fender | PanicMan: Have you even written your own dialplan yet? |
| 03:50.14 | panicman | not yet |
| 03:50.30 | [TK]D-Fender | PanicMan: Go write an AGI script or something to pick the number to set and just go do it |
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| 03:50.44 | panicman | ok sir |
| 03:50.50 | [TK]D-Fender | PanicMan: If you haven't written your dialplan then you haven't even started with * yet |
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| 04:13.12 | alvariux | hello |
| 04:14.13 | alvariux | i have two two AEX800 cards, when we are making about 10 simultaneous calls we have echo or static problems, any advice can you give? |
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| 05:15.35 | cvnet | anyone here willing to help me setup a ipsec on Fedora, the provider is asking to send sip via tunnel? I'm willing to pay your help |
| 05:19.12 | xorl | so, what package holds enter-ext-of-person or what was it replaced by |
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| 05:19.40 | luckyaba | anyone know a way to find out what RTP port a phone is using when receiving a call from an external number? |
| 05:24.12 | *** join/#asterisk Severian (n=ralph@ppp-70-251-105-18.dsl.rcsntx.swbell.net) |
| 05:26.08 | Severian | My impression from what I can find online is that IPV6 support in Asterisk won't come soon. Can I get a comment on that? A project I am helping with is considering moving to ipv6 and I think that will be a problem. This is a mesh voip application for third world countries. |
| 05:34.00 | denon | Severian: texas is pretty rural, but "third world" seems like an exaggeration |
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| 05:42.49 | Severian | denon, well I suppose I will test some in Texas. The project is run by a guy somewhere in Australia. I don't know if it is rural enough for you there, but the first real implementation is for villages in South Africa. |
| 05:45.12 | denon | was just kidding .. |
| 05:45.13 | xorl | hmmm so I have teliax SIP line routing through my * box |
| 05:45.22 | denon | I think 1.6 is likely to see ipv6.. |
| 05:45.34 | denon | and no, Ive been to AU .. its not all real rural :) |
| 05:45.35 | xorl | http://pastebin.com/d4a97077f i have that, if I change the Dial command to correspond to the uelephant instead of the '666' it fails |
| 05:45.45 | xorl | bah it doesn't fail, it fails with exten 666 |
| 05:46.59 | xorl | n/m i'm a retard |
| 05:47.06 | xorl | n != answering 1 = answering haha |
| 05:47.10 | xorl | fixed it. |
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| 05:49.31 | Nugget | heh |
| 05:50.20 | xorl | Didn't even see my typo eff up |
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| 05:57.58 | edibrac | is there a site out there that has recordings of common phone noise problems and what they indicate? |
| 05:58.38 | edibrac | the one I have, i recorded and played back to a AT&T guy I saw working on the street. He said it there could be ground noise that could have basically brought down our PRI. |
| 05:58.39 | Nugget | no, because it's way more fun when people try to describe a noise on irc. |
| 05:59.06 | edibrac | lol yeah - my noise goes like: "kruugm zzim blaaaa anewt" |
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| 05:59.49 | edibrac | i know the "iphone-is-too-close" nose |
| 05:59.56 | Nugget | huh, weird. dunno. I only know what it means when the noise is like "kruugm zzim blaaaa amoooooeual" |
| 06:00.09 | edibrac | noise. but this one I was having was quite different. and everyone was reproting. |
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| 06:13.23 | blaxthos | anyone ninja with cisco 7960 ? |
| 06:13.45 | blaxthos | phone pulling proper config, but won't register with trixbox |
| 06:13.55 | blaxthos | problem appeared after i tried to update firmware |
| 06:14.25 | blaxthos | well, after i tried to upgrade and it got stuck in reboot loop, so i downgraded |
| 06:14.30 | blaxthos | was working previously :( |
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| 07:29.02 | chodorenko | Hi All |
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| 07:31.58 | chodorenko | It is necessary to accept an order of 500 simultaneous calls |
| 07:31.58 | chodorenko | All of them should be written down, has written a small script of testing which does ring connection Dial Local/8888, and in 8888 call AGI script which lies causes DIAL on 8888 |
| 07:31.59 | chodorenko | I start to test, all is normal till certain time, and then it is literally for pair calls (гдеÑо with 210 on 250 call) value of parametre of loading SYSTEM (on top it is visible) sharply increases, and from power the CPU not especially depends |
| 07:31.59 | chodorenko | Without record 500 calls it is fulfilled normally |
| 07:32.00 | chodorenko | In what format there is a record especially does not influence |
| 07:32.22 | chodorenko | howe it`s fix ? |
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| 08:52.19 | tompaw | Hi. |
| 08:52.32 | tompaw | Is it now possible to define an ip range for sip peer? |
| 08:52.39 | tompaw | Or does it still have to be one peer per ip? |
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| 09:46.43 | iksik | tompaw, I've got peer with host dynamic, and it works :P |
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| 10:34.26 | nextime | hello, asterisk 1.6, chan_dadhi for isdn BRI, sip extensions for snom phones. I have a snom 320 from wich i watch other phone status by using exten => XXX,hint,SIP/${EXTEN} in my dialplan and subscribing to watch the presence status from the snom 320 phone |
| 10:34.48 | nextime | how can i do the same thing to watch not only other phones, but also my quad-bri channels status? |
| 10:35.39 | nextime | something like hint,DAHDI/1 |
| 10:35.52 | nextime | is there anything i can use to obtain this result? |
| 10:47.54 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:48.43 | pif | hi, can I use native kernel mISDN with asterisk 1.6 ? |
| 10:49.18 | pif | more generally, what option do I have with a hfcpci card with asterisk 1.6? |
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| 11:21.07 | wonderworld | on a PRI every 5th call or so terminates with ISDN Cause Code 16 "normal call clearing". why would that happen? |
| 11:29.07 | pif | tzafrir_laptop: hi? |
| 11:33.24 | X-Rob_ | wonderworld, it means the call dropout has nothing to do with the PRI |
| 11:33.44 | X-Rob_ | the PRI is being told to end the call for some reason. If it's outbound calls, look above there in the logs for a reason |
| 11:34.53 | wonderworld | so the problem must be local in my asterisk? |
| 11:38.54 | thomas | hello. |
| 11:39.30 | thomas | I would like when I dial "00891234556" then only 01234556 << without 089 |
| 11:39.32 | thomas | is it posobiel? |
| 11:43.10 | wonderworld | X-Rob_: what logfile should be checked? on the console with verbosity 99 i simply get the dial-command and afterwards ich NoOp out the Hangupcause. There seems to be no info in between |
| 11:43.48 | X-Rob_ | /var/log/asterisk/full |
| 11:44.39 | beek | thomas: yes |
| 11:45.04 | kaldemar | thomas: ${EXTEN:3} will cut first three digits |
| 11:46.53 | thomas | beek / kaldemar: yes, i mean i need a gotoif or what? |
| 11:46.58 | leifmadsen | More videos coming, but here is the first set from IT360, my presentation on "Unusual Uses: What you didn't know your Asterisk system could do." Available from http://leifmadsen.wordpress.com/2009/06/09/leif-madsen-it360-april-2009/ |
| 11:47.39 | leifmadsen | thomas: exten => __X.,1,Verbose(1,Here is the extension with the first 3 numbers cut off: ${EXTEN:3}) |
| 11:48.11 | thomas | leifmadsen: hello. yes i mean when i have to begin from my number "0089" then cut "089" when not then not. |
| 11:48.14 | leifmadsen | don't copy paste that, because I think I accidently pressed _ (underscore) twice there, so be sure to remove the first underscore. |
| 11:48.35 | thomas | have not: exten => _0.,n,Dial(SIP/${ziel}) |
| 11:49.40 | thomas | or i need exten => _0089.,n,Dial(SIP/${EXTEN:4}) ? |
| 11:49.50 | thomas | äh ${EXTEN:3} |
| 11:49.57 | leifmadsen | I don't know what you're asking now |
| 11:50.12 | leifmadsen | yes, what you said |
| 11:50.24 | thomas | leifmadsen: i need both lines or? |
| 11:50.31 | leifmadsen | yes |
| 11:50.41 | leifmadsen | _0089. is more specific of a match over _0. |
| 11:50.59 | X-Rob_ | thomas: EXTEN = 123456789 ... ${EXTEN:1}=23456789.. ${EXTEN:5}=6789 |
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| 11:51.03 | leifmadsen | 0999 will match the _0., and 008912345 will match _0089. |
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| 11:51.28 | leifmadsen | ${EXTEN:offset:length} |
| 11:51.37 | leifmadsen | that's the format of channel variable manipulation |
| 11:51.44 | thomas | ${EXTEN:offset:length} hm? ok |
| 11:51.53 | leifmadsen | the 2nd value is not required |
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| 11:52.06 | leifmadsen | leaving it out will mean use the rest of the string |
| 11:52.28 | leifmadsen | EXTEN:2:3 = 345 |
| 11:52.35 | leifmadsen | where EXTEN = 123456789 |
| 11:52.58 | thomas | emm |
| 11:53.24 | leifmadsen | just ignore that unless you need it |
| 11:53.53 | thomas | leifmadsen: when i dial: 008912345678 and I would like only the 012345678 |
| 11:54.01 | thomas | then |
| 11:54.02 | thomas | hm |
| 11:54.48 | leifmadsen | exten => _0089.,1,Set(NUMBER=0${EXTEN:4}) |
| 11:54.48 | thomas | oh ok |
| 11:54.48 | thomas | its easy |
| 11:54.48 | leifmadsen | take off the first 4, add the 0 |
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| 12:02.45 | thomas | leifmadsen: okay, thank you very much |
| 12:03.06 | tzafrir_laptop | pif, a bit busy now... |
| 12:03.08 | beek | leifmadsen: Nice presentation. Thanks for posting it. |
| 12:03.26 | leifmadsen | beek: np! hope it turned out ok -- I haven't even watched it yet :) |
| 12:04.16 | beek | It's really nice in that it automatically chains from one part to another. That's the first I've seen YouTube do that on a multipart video |
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| 12:06.35 | beek | morning jaytee |
| 12:06.40 | leifmadsen | beek: oh, that's just because I built a playlist and linked to the playlist :) |
| 12:06.42 | jaytee | morning beek |
| 12:06.56 | beek | leifmadsen: you're the first I've seen do that. |
| 12:06.58 | leifmadsen | beek: apparently I'm a youtube guru (I just started using it yesterday :)) |
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| 12:07.07 | leifmadsen | crazy |
| 12:07.15 | leifmadsen | coppice: I'm talkin' to you |
| 12:08.10 | leifmadsen | beek: now to down sample the rest of the presentations from the 2.5 GB's each and get them sent over to my video partner to post all of them. That was really just me playing around with YouTube |
| 12:08.17 | leifmadsen | the 10 minute limit is slightly annoying though... |
| 12:09.00 | beek | leifmadsen: You've inspired me to try IMAP voicemail again. The last time I used it it had some issues. I think in 1.6.0.10 I can poll to see if the voicemail email had been "read" to turn off the message waiting lamp. |
| 12:10.39 | dandre | Hello, |
| 12:11.26 | leifmadsen | beek: yes, I believe so. I haven't used it in a bit now, I just played around with it in a lab setting -- I hope to get a chance to configure it more in a "production" use and document that as well |
| 12:11.41 | leifmadsen | all of the stuff in that presentation was more "lab" quality, and not production quality |
| 12:12.48 | beek | leifmadsen: I'm trying to remember if it was the 1.6.1 series that added that polling feature or 1.6.0.x somewhere. I need to go back and read the change logs to find out. |
| 12:13.05 | leifmadsen | beek: ya, I'm not 100% sure which version added it either |
| 12:13.09 | leifmadsen | dandre: hello! |
| 12:13.20 | dandre | I am looking for a solution that consist for one particuliar phone to some sort of automatic dialing. That is when a phone goes affhook, a number is automatically dialed. I don't know how to do that (in extensions.conf, finding some specific phone, a sip phone or an analog phone? ... |
| 12:13.25 | coppice | leifmadsen: you can say anything you like when I'm not connected |
| 12:14.00 | Aiatek | in analog phone |
| 12:14.15 | Aiatek | you can do that in chan_dahdi.conf |
| 12:14.24 | Aiatek | <dandre> |
| 12:14.35 | dandre | I still use chan_zap |
| 12:14.46 | Aiatek | in zapata.conf |
| 12:15.00 | Aiatek | use immediate=yes |
| 12:15.23 | Aiatek | and the phone will go to the s extension of the context that belong |
| 12:15.50 | Aiatek | if the phone is in context=users |
| 12:16.24 | Aiatek | in extension.conf in context users the phone will execute the s extension in that context |
| 12:18.49 | beek | leifmadsen: Are you going to post the dialplan/slides somewhere? The dialplan around the calendar is illegible. |
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| 12:20.12 | dandre | Ok thanks Aiatek |
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| 12:20.38 | leifmadsen | beek: www.leifmadsen.com/presentations.php |
| 12:20.49 | leifmadsen | I should add a notation for that |
| 12:20.54 | beek | leifmadsen: thanks! |
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| 12:24.27 | leifmadsen | beek: added, thanks! |
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| 12:32.47 | jaytee | leifmadsen, wow! that's pretty cool! I'm gonna look at the CURL stuff later when I get some time |
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| 12:33.51 | leifmadsen | jaytee: ya, I found it really neat |
| 12:34.02 | leifmadsen | jaytee: and this comes from someone who is very much a func_odbc person :) |
| 12:34.09 | leifmadsen | s/person/advocate/ |
| 12:35.27 | afink | Morning everyone, I recently upgraded from zaptel to the latest stable version of dahdi. Now my HPEC has stopped working. Anyone run into this? or have any suggestions? |
| 12:35.29 | jaytee | I kinda like that fact that you list your consulting fees right up front on your page. alot of people are kind of guarded and don't let people shopping for services know up front what they're getting into. usually a "well, it depends on the job" |
| 12:38.58 | leifmadsen | jaytee: yep, it also is a good filter for someone looking for a $20/hr consultant :) |
| 12:39.49 | afink | Anyone have HPEC running with the latest stable version of dahdi on a TDM800P? |
| 12:39.54 | coppice | you can make a good business sub-contracting that to people at $2 an hour |
| 12:40.21 | jaytee | afink, I don't use HPEC, I use hardware EC but there are tons of Google hits on that issue. I'm pretty sure there are config changes that need to be done and IIRC some modifications to the source before compiling dahdi |
| 12:42.13 | coppice | Digium's HPEC doesn't work out of the box with Digium's DADHI? funky :-\ |
| 12:42.25 | afink | ok, jaytee I think I have made the proper config changes, everything starts no errors but no echo cancellation. I will check out the source modifications to see if I missed something |
| 12:44.45 | jaytee | afink, the zaptel version of HPEC checked for the existence of /dev/zap/ctl so you may need to upgrade the version of HPEC |
| 12:45.32 | jaytee | check the readme file |
| 12:45.39 | afink | jaytee: I did that with digium's help |
| 12:45.50 | jaytee | and still no EC? |
| 12:45.54 | afink | nope |
| 12:45.58 | jaytee | wow |
| 12:46.10 | jaytee | that's why I love hardware EC :-) |
| 12:47.08 | afink | jaytee: I think I would have a lot less grief if I got some hardware ec |
| 12:47.50 | [TK]D-Fender | afink: pastebin your configs |
| 12:48.02 | afink | [TK]D-Fender: which ones would you like to see? |
| 12:48.10 | coppice | afink: hardware EC gives people plenty of grief too |
| 12:48.18 | [TK]D-Fender | afink: yor DAHDI configs |
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| 12:50.52 | afink | '/etc/dahdi/system.conf http://pastebin.com/m4174d482 |
| 12:51.54 | afink | dahdi-channels.conf : http://pastebin.com/m283c0efa |
| 12:52.45 | [TK]D-Fender | afink: And "dahdi_cfg -vvvv" |
| 12:53.26 | [TK]D-Fender | afink: And where is chan_dahdi.conf? |
| 12:53.42 | afink | dahdi_cfg: http://pastebin.com/m31ec4443 |
| 12:54.55 | pif | tzafrir_laptop: in dadhi 1:2.2.0~dfsg~rc5-1 the zaphfc module crashes linux |
| 12:55.17 | afink | [TK]D-Fender: chan_dahdi.conf http://pastebin.com/m474b0867 |
| 12:56.25 | [TK]D-Fender | afink: I don't see that inluding "dahdi-channels.conf" that you showed earlier... |
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| 12:57.04 | bobsaccamano | hi..im using cisco 7940 ip phones as sip peers in asterisk 1.6.0.1..im getting only intermittent sound |
| 12:57.12 | bobsaccamano | and after sometime the sound vanishes |
| 12:57.22 | bobsaccamano | can anyone help?? |
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| 12:58.19 | afink | [TK]D-Fender: not sure but does it have anything to do with using GUI? =| |
| 12:58.31 | [TK]D-Fender | ....... |
| 12:58.47 | [TK]D-Fender | gah |
| 12:59.36 | afink | I'm trying to get away from it, it is more of a restriction than anything at this point |
| 13:01.30 | tzafrir_laptop | pif, the open bug on the debian package? |
| 13:03.01 | m-i-l-a-n | hi all! i want to setup an automated email2fax system. hardware is coming next week, im wondering AsteriskNOW any good for that? |
| 13:03.46 | [TK]D-Fender | m-i-l-a-n: taht is jsut a distro with * isntalled |
| 13:03.59 | [TK]D-Fender | m-i-l-a-n: All the things you need are THIRD PARTY |
| 13:04.01 | m-i-l-a-n | i know, want to have a dedicated machine |
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| 13:04.20 | m-i-l-a-n | oh. so distro doesnt matter then |
| 13:04.23 | leifmadsen | [TK]D-Fender: really? third party needed to be capitalized? |
| 13:04.27 | [TK]D-Fender | m-i-l-a-n: * is just one little piece of the puzzle |
| 13:04.30 | pif | tzafrir_laptop: yes, that's the one. Should I use vzaphfc or zaphfc for a hfc-pci card? |
| 13:05.02 | [TK]D-Fender | leifmadsen: YES :) Helps prevent people from beating down asterisk.org & Digium.com for software they don't make :) |
| 13:05.03 | m-i-l-a-n | i plan to use hylafax and iaxmodem if thats not overkill or dead end |
| 13:05.18 | [TK]D-Fender | m-i-l-a-n: No, those are the bits you should be looking at |
| 13:05.56 | [TK]D-Fender | m-i-l-a-n: * may simply provide access to your PSTN interface for it. |
| 13:08.18 | m-i-l-a-n | also, im still uncertain if it will work, ive been trying to read up on how to send faxes to local lines through sip. a real PITA |
| 13:08.55 | coppice | FAX and SIP go together like George Bush and a brain |
| 13:09.05 | m-i-l-a-n | ;( |
| 13:09.20 | m-i-l-a-n | alternative is me sending fax by hand each time |
| 13:09.38 | m-i-l-a-n | which i cba doing after holidays ;) |
| 13:09.51 | coppice | the sending mechanism is not the issue. the medium is |
| 13:10.34 | m-i-l-a-n | i know. its about ~40 target numbers. all have no internet only local lines phone and fax |
| 13:11.02 | m-i-l-a-n | hate it that they just wont set up their business properly |
| 13:11.08 | [TK]D-Fender | m-i-l-a-n: What is this about sending by hand? You have regular lines, jsut use them |
| 13:11.41 | Aiatek | <m-i-l-a-n> if you dont know about setting VoIP faxes there are pre-configured distros with fax support |
| 13:11.59 | Aiatek | you can used it just for fax like a dedicated fax machine |
| 13:12.15 | m-i-l-a-n | Aiatek: thats interesting! i will read up;) |
| 13:12.51 | m-i-l-a-n | [TK]D-Fender: problem is. i dont have local line. only the stores have, ive got voip only |
| 13:13.11 | [TK]D-Fender | m-lOh well |
| 13:13.40 | m-i-l-a-n | as far as i understood it, this might be a problem. maybe it isnt at all. i'll see |
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| 13:46.01 | Zhad | For some reason, I can't seem to get call announcements working |
| 13:46.10 | Zhad | s/call/queue/; |
| 13:46.20 | Zhad | more irritatingly, I don't even know when they stopped working. |
| 13:49.37 | Zhad | http://pastebin.com/d1ec67221 |
| 13:49.54 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 13:50.28 | Zhad | It's worth noting, that when the calls come in, most of the time there are agents available, just not answering. |
| 13:50.59 | Zhad | hence the small timeout and retry |
| 13:51.11 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:51.22 | Zhad | There is also no announcement on joining the queue. |
| 13:53.37 | Zhad | Dammit, forgot to mention, this is with 1.6.0.9 |
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| 14:04.14 | [TK]D-Fender | Zhad: It looks like your queu only rings people for 15 seconds, which is the periodic message interval. This means Queue should exit and they will never hear it |
| 14:04.46 | Zhad | according to the documentation on voip-info, If I have the timeout later than that, the announcement will move until the timeout period |
| 14:04.53 | Zhad | because there are phones ringing. |
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| 14:05.20 | Zhad | the way round it is suppposed to be to set the timeout to the accounce period and let it retry. |
| 14:05.24 | Zhad | which doesn't work either. |
| 14:05.50 | Zhad | although voip-info could be wrong, it wouldn't be the first time. |
| 14:06.20 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:06.20 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
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| 14:09.25 | [TK]D-Fender | Zhad: In your dialplan you seem to indicate a maximum Queue wait time of 15 seconds. not jsut an agent ring time, but max time in Queue |
| 14:09.26 | Zhad | changing timeout doesn't fix it (worth a shot though). |
| 14:10.03 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.82) |
| 14:10.31 | Zhad | That was believe it or not, added to try and fix it. |
| 14:10.54 | Zhad | changing this to 60 seconds, and I still don't get an announcement. |
| 14:11.04 | Zhad | neither do I if I remove it entirely. |
| 14:11.13 | [TK]D-Fender | Zhad: disable the standard announce first along with the dialplan based timeout, and pastebin new configs and a failed call |
| 14:11.29 | [TK]D-Fender | Zhad: And of course show us you have that recording |
| 14:11.30 | *** join/#asterisk xpot-mobile (n=james@static-66-182-88-85.bbsc.net) |
| 14:12.21 | Zhad | I don't have an announce set. |
| 14:12.40 | [TK]D-Fender | ? |
| 14:12.54 | [TK]D-Fender | Zhad: Please clarify... |
| 14:14.11 | Zhad | no announce= line. |
| 14:14.41 | Zhad | unless you meant disable the playback bit. |
| 14:14.45 | Zhad | in the dialplan. |
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| 14:20.47 | Zhad | http://pastebin.com/d36264809 |
| 14:21.10 | Zhad | The call doesn't fail so much as the music continually plays (unless someone answers) and there are no call announcements. |
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| 14:24.36 | [TK]D-Fender | Zhad: Ah, you commented out the timeout in your queues.conf definition. |
| 14:24.40 | [TK]D-Fender | Zhad: You need that.. |
| 14:25.17 | Zhad | when I make changes, is there a way of reloading the config without restarting asterisk? |
| 14:25.26 | [TK]D-Fender | Zhad: "reload" |
| 14:25.49 | Zhad | queues doesn't have a reload |
| 14:25.53 | Zhad | only on rules |
| 14:26.26 | [TK]D-Fender | Zhad: "reload" will take queues.conf changes |
| 14:27.06 | Zhad | thanks, working now |
| 14:27.22 | Zhad | about 2 seconds after the timeout appears in the console (which is odd, but not a problem). |
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| 14:28.27 | Zhad | Is there a way to make the announcement come on when someone joins the queue as well? |
| 14:29.10 | [TK]D-Fender | Zhad: Nope. |
| 14:29.57 | [TK]D-Fender | Zhad: Announcements only play between calling agents and typically when you join it starts ringing. |
| 14:30.19 | Zhad | fair enough |
| 14:30.21 | [TK]D-Fender | Zhad: You could play one before entering the queue but that won't let an agent interrupt it |
| 14:30.37 | *** join/#asterisk meingbg (n=user@173-45-238-108.slicehost.net) |
| 14:30.40 | [TK]D-Fender | Zhad: And delays your client being answered as quickly as might be possible |
| 14:31.06 | *** join/#asterisk devyll (n=email@89.36.24.2) |
| 14:31.52 | devyll | can I set the queue in such way that any caller on hold can choose to leave a voice message instead of continue holding ? |
| 14:32.20 | *** join/#asterisk VoipForces (n=kvirc@office.privalodc.com) |
| 14:32.43 | Zhad | can you do that using the context option in queue? (although that probably only works when you are queueing). |
| 14:32.56 | VoipForces | Anyone knows something similar to FOP but would be able to list active channels as well as support up to 2000 estensions ? |
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| 14:33.33 | meingbg | Hi, we are currently considering to get our own asterisk server going to cut expenses on IP telephony... What are some of the main aspects I should go through before making the decision? |
| 14:34.27 | Zhad | mostly the same as buying any new kit, the first consideration is who is going to support it. |
| 14:35.58 | meingbg | Zhad: Support it, as in making sure it works all the time? I guess I'm the one who's supposed to do that. |
| 14:36.03 | [TK]D-Fender | Zhad: No way to get the announce first. the context is an exit option only |
| 14:36.25 | [TK]D-Fender | meingbg: * has nothing to do with cutting expenses of IP telephony. |
| 14:36.52 | Zhad | TK> I was referring to devyll's problem. |
| 14:37.02 | [TK]D-Fender | meingbg: Gas at the pump costs the same for a Honda as it does an Toyota |
| 14:38.07 | Zhad | If all you want to achive is to be able to make voip calls, then there is probably something you can do with your existing kit. |
| 14:39.21 | errr | [TK]D-Fender: it sure did cut our costs... we were leasing a crappy expensive pbx, now we manage our own with asterisk which cut the costs of our pbx by about 30k a year |
| 14:39.31 | *** join/#asterisk feder (n=feder@host.190.15.192.172.static.itcsa.net) |
| 14:39.34 | meingbg | [TK]D-Fender: Zhad: Well, it's not really about the cost for phone calls. Currently we're buying IP telephone as a service, and we pay monthly for everything; queues, Voice mail, everything. |
| 14:40.07 | feder | hello, is there a way to do inserts through realtime? |
| 14:40.34 | meingbg | I was wondering if running our own * server would cut those costs, leaving us with just the costs for the actual phone calls and whatever it takes to keep the server running. |
| 14:40.47 | [TK]D-Fender | meingbg: Well if you can find a cheaper service you can rely on, then * can do those other things |
| 14:42.04 | meingbg | [TK]D-Fender: Sounds good. BTW, why is somebody offering this for free? |
| 14:43.04 | [TK]D-Fender | meingbg: Offering what for free? |
| 14:43.04 | meingbg | The server software. |
| 14:43.04 | [TK]D-Fender | meingbg: Go ask that of OpenOffice.org, etc. |
| 14:43.08 | meingbg | [TK]D-Fender: I'd like to. |
| 14:43.08 | [TK]D-Fender | meingbg: Digium sells CARDS because of it |
| 14:44.49 | Zhad | If you buy the business version, you get a support contract included. |
| 14:45.06 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 14:45.11 | meingbg | [TK]D-Fender: Can I get an * feature list from somewhere, so I can tell if it'll cover our needs? |
| 14:45.15 | *** join/#asterisk cc6 (n=cc6@pool-72-91-71-74.tampfl.fios.verizon.net) |
| 14:45.31 | pta200 | anybody know of any potential issues using dundi across a group of servers some of which are running 1.2 and others 1.4? |
| 14:45.51 | meingbg | Zhad: Sounds fair enough. Are corporations still allowed to use the free version? |
| 14:45.53 | [TK]D-Fender | meingbg: asterisk.org |
| 14:46.13 | [TK]D-Fender | meingbg: and yes, anyone can |
| 14:46.22 | [TK]D-Fender | meingbg: GPL 2 |
| 14:47.25 | Zhad | didn't know it was GPL 2, I thought it was something more perverse like mozilla. |
| 14:47.49 | *** join/#asterisk dacs (n=dacs@unaffiliated/dacs) |
| 14:48.23 | coppice | its not GPL2. it just pretends to be. A GPL2 project can accept GPL2 code. Asterisk will not |
| 14:49.11 | dacs | howdy folks |
| 14:49.30 | *** join/#asterisk wierdo (n=jimmy@wifi-traf5.networx-bg.com) |
| 14:49.47 | dacs | can i please know if anyone using asterisk with Debian Lenny please? if so where and how can i get it |
| 14:49.51 | dacs | thanks in advance |
| 14:50.32 | *** part/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 14:51.12 | edibrac | can't you apt-get install asterisk on debian? |
| 14:52.07 | [TK]D-Fender | coppice: Maybe the people who consider themselves "mainline contributors", no... however you are free to fork. |
| 14:52.18 | dacs | edibrac: i will install 1.4 if i did that |
| 14:52.28 | [TK]D-Fender | coppice: Its a question of who considers themselves the "original" or "official" branch |
| 14:52.38 | mmlj4 | dacs: then try asterisk.org |
| 14:53.44 | mmlj4 | could be wose, you could be running etch, Asterisk 1.2.13 |
| 14:57.31 | pta200 | So no one has run into any issues running dundi between asterisk 1.2 and 1.4 servers? |
| 14:57.35 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 15:02.22 | rue_mohr | would any setting anyone can think of mess up dtmf on a bridged dahdi card |
| 15:02.34 | rue_mohr | as in fxo to fxs call |
| 15:02.37 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 15:03.09 | [TK]D-Fender | rue_mohr: Gains distoring the tones, EC freakout, etc |
| 15:03.44 | [TK]D-Fender | rue_mohr: I've replaced several Digium cards my clients had that were jsut generally crackly as well |
| 15:03.57 | rue_mohr | but you still insist that the only way it should be adjusted is by its default 0db values, right? |
| 15:04.18 | [TK]D-Fender | rue_mohr: Are you still running psycho gains? |
| 15:04.28 | rue_mohr | no just insane |
| 15:04.30 | Zhad | wouldn't argue with coppice' forking. |
| 15:05.02 | jameswf | huh huh heh heh he said forking |
| 15:05.10 | rue_mohr | see, I'd expect crackle if its gain was too high |
| 15:07.30 | rue_mohr | right now I'm 8/-10 |
| 15:08.12 | jameswf | rue_mohr: at some point it is okay to call the telco and tell em their lines are crap |
| 15:08.55 | rue_mohr | its fine with standard phones |
| 15:09.16 | rue_mohr | but I have no way of knowing what the actaul pots levels are |
| 15:09.26 | jameswf | rue_mohr: gains should be 0, if 0 is no good fxotune, if still bad then start adjusting gains |
| 15:09.32 | rue_mohr | I have no idea is the level off the sip phones is ok |
| 15:10.00 | Zhad | don't you just hate it when you go through a checkout process on a website and instead of getting a confirmation screen you get a browser crash. |
| 15:10.23 | rue_mohr | jameswf, at 0, the people in the office cant hear the far party |
| 15:10.51 | jameswf | rue_mohr: did you run fxotune... |
| 15:11.44 | jameswf | rue_mohr: also you can see if the audio drop is @the card or @ the sip phone with ztmonitor |
| 15:12.06 | rue_mohr | yes, fxotune came up with 6,0,0,0,0,0,0 for all the channels |
| 15:12.06 | [TK]D-Fender | jameswf: Are you kidding? there's nothing he hasn't done with that car after all these months. |
| 15:12.08 | [TK]D-Fender | the SLUT :p |
| 15:12.23 | rue_mohr | its not ztmonitior anymore, its dahdi_monitor |
| 15:12.46 | jameswf | I don;t monitor my dahdi he is an A** hole |
| 15:14.16 | rue_mohr | the dahdi monitor is almost useless without a db reference |
| 15:14.35 | jameswf | I am waiting to beat the dead horse again... someone calls and says I have dahdi issues, I can reply well go get counseling.. |
| 15:14.35 | rue_mohr | -vv... |
| 15:14.48 | *** part/#asterisk pta200 (n=paolo@69.38.173.162) |
| 15:16.01 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:16.38 | *** part/#asterisk mattbUK (n=mattbrid@82-46-93-20.cable.ubr16.stav.blueyonder.co.uk) |
| 15:18.46 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-abee0a99acd8cc1b) |
| 15:19.00 | rue_mohr | so dont you agree that 1mw shoudl be 0db on the pots, 0db on rtp ? |
| 15:19.05 | *** join/#asterisk Juggie (n=Juggie@CPE001601df17fb-CM001ceac25ada.cpe.net.cable.rogers.com) |
| 15:20.01 | rue_mohr | I have a book with info on what the system return gain should be (max) but dont have it |
| 15:21.17 | rue_mohr | haha sip show inuse dosn't work |
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| 15:24.03 | *** join/#asterisk moy (n=moy@74.12.123.90) |
| 15:27.24 | rue_mohr | why dosn't sip show inuse work!? |
| 15:28.43 | jaytee | "works for me!" |
| 15:28.54 | rue_mohr | 1.4.22 |
| 15:29.12 | rue_mohr | known issues? |
| 15:29.27 | rue_mohr | I'm gonna upgrade it to .25.1 I just need downtime |
| 15:29.46 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.175.58) |
| 15:30.53 | *** join/#asterisk ramindia (n=Administ@202.63.96.10) |
| 15:31.01 | carrar | woah |
| 15:31.04 | carrar | WORKS for me too! |
| 15:31.26 | rue_mohr | I have someone on the phone behind me and it says nothing |
| 15:31.47 | bmoraca | do you have call limits set on your sip peers? |
| 15:31.56 | rue_mohr | whats a call limit |
| 15:31.58 | carrar | no |
| 15:32.07 | bmoraca | call-limit |
| 15:32.08 | carrar | call limits is so last year |
| 15:32.12 | *** join/#asterisk hfb (n=hfb@pool-96-247-109-136.lsanca.dsl-w.verizon.net) |
| 15:32.16 | rue_mohr | why would you limit calls? |
| 15:32.30 | bmoraca | certain queue features didn't work without them |
| 15:32.34 | rue_mohr | we hire centipedes for a reason |
| 15:32.46 | kaldemar | for people who can't multitask |
| 15:32.52 | Zhad | wasn't it usually the fault of the SIP client not reporting state properly? |
| 15:33.12 | rue_mohr | is this related to my users losing calls when they put more than one on hold at a time? |
| 15:33.22 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 15:33.31 | rue_mohr | well these are polycom phones, I dont think they do anything right |
| 15:33.44 | carrar | I think I still use that for blf crap |
| 15:33.50 | bmoraca | rue_mohr: that's most likely a layer 8 issue |
| 15:33.57 | rue_mohr | but my blf works fine |
| 15:33.57 | carrar | been a while |
| 15:34.44 | rue_mohr | 8 is well into application isn't it? |
| 15:34.52 | bmoraca | 8 is the user layer |
| 15:34.55 | [TK]D-Fender | :D |
| 15:34.58 | rue_mohr | :) |
| 15:35.17 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-e6495d3438b14c87) |
| 15:35.18 | rue_mohr | oh the multiple hold thing, yea, definitly 8 |
| 15:35.40 | carrar | malloc errors! |
| 15:35.51 | rue_mohr | no interface problems |
| 15:35.52 | [TK]D-Fender | rue_mohr: http://tinyurl.com/496svm |
| 15:36.30 | carrar | maybe his is that one dry container |
| 15:36.33 | rue_mohr | thats how you unload logs in canada |
| 15:36.59 | rue_mohr | but logs float |
| 15:37.22 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 15:37.52 | rue_mohr | ok, back to it, tell me please, if I set a limit, will more than the limit of calls be able to get to the phone at once? |
| 15:38.02 | rue_mohr | or is it "busy" after the limit |
| 15:39.05 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 15:39.23 | bmoraca | rue_mohr: it'll come back as unavailable. however, most people just set the limit exceptionally high...such as 99 |
| 15:39.25 | kaldemar | FYI, call-limit is deprecated in favor of group functions. |
| 15:39.42 | bmoraca | i don't think that's the case in 1.4.22... |
| 15:40.22 | rue_mohr | I had all the gains on the dahdi (fxo and fxs) set to rxgain=8 txgain=-10 so for a call between dahdi channels, that would be -2db both ways, right? |
| 15:40.43 | rue_mohr | ah, so for sip show inuse to work, the limit just has to be set to SOMETHING |
| 15:40.44 | rue_mohr | ? |
| 15:40.53 | kaldemar | it was deprecated in 1.6. should work though. but, may pay off to do limitations with group functions already. |
| 15:41.17 | rue_mohr | I just was sip show inuse to work |
| 15:41.25 | rue_mohr | ? |
| 15:42.25 | rue_mohr | hmm nobody knows... |
| 15:42.32 | kaldemar | knows what? |
| 15:42.37 | bmoraca | why not give it a try? |
| 15:42.47 | rue_mohr | cause its a live systemt hat inuse |
| 15:43.39 | bmoraca | so? reload doesn't impact current or future calls. set it on your extension, reload, and see what happens |
| 15:43.47 | [TK]D-Fender | rue_mohr: So? These settings don't hurt anything. |
| 15:44.23 | [TK]D-Fender | rue_mohr: and I've never really ever heard of "sip show inuse", nor can I see a need for it |
| 15:44.30 | *** join/#asterisk csiadmin (n=csiadmin@81.144.152.52) |
| 15:44.32 | [TK]D-Fender | rue_mohr: Other ways to see whats going on |
| 15:45.19 | *** join/#asterisk [netman] (n=netman@118.Red-83-50-81.dynamicIP.rima-tde.net) |
| 15:45.56 | csiadmin | hi everyone |
| 15:46.13 | eppigy | hello |
| 15:46.52 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:46.55 | rue_mohr | I have to shut down and restart asterisk for gain changes to take effect, that tends to interrupt calls in progress |
| 15:47.09 | rue_mohr | I cant tell if there are any calls in progress cause sip show inuse dosn't work |
| 15:47.14 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 15:47.25 | [TK]D-Fender | rue_mohr: For gains, yes, everything else, no |
| 15:47.39 | csiadmin | Anyone help me get an IRQ that is not shared for a digium card? I've moved it to every slot and even disabled USB |
| 15:47.40 | rue_mohr | can you tell me how to know if there are calls going on without using sip show inuse |
| 15:47.41 | [TK]D-Fender | rue_mohr: "core show channels concise" <- |
| 15:47.53 | [TK]D-Fender | rue_mohr: Who says your calls have to be SIP? |
| 15:47.55 | Zhad | sip show channels |
| 15:47.59 | Zhad | core show channels |
| 15:48.03 | [TK]D-Fender | rue_mohr: You've insaned </katty> |
| 15:48.27 | Zhad | sip show inuse is awfully verbose (tells you everything that's not inuse as well). |
| 15:48.34 | Corydon76-dig | csiadmin: which card? |
| 15:48.39 | rue_mohr | no it says nothing |
| 15:48.40 | Zhad | It mostly depends on what's asking it really. |
| 15:48.53 | csiadmin | Corydon76-dig, it's a te121 |
| 15:49.11 | rue_mohr | core show channels consise says nothing |
| 15:49.32 | rue_mohr | sip show channels says nothing |
| 15:49.34 | Zhad | on mine it gives a line for every defined user and lists details, then every defined peer and lists details. |
| 15:49.35 | [TK]D-Fender | rue_mohr: then there are no calls |
| 15:49.49 | rue_mohr | ah let me create one for sure and test again |
| 15:49.51 | Corydon76-dig | csiadmin: there is no need to isolate the interrupt for that card |
| 15:49.53 | Zhad | (I was refereing to sip show inuse). |
| 15:50.17 | rue_mohr | ah, k |
| 15:50.38 | Corydon76-dig | csiadmin: VoiceBus doesn't suffer when a card misses an interrupt, which is what isolation was supposed to help with |
| 15:50.55 | *** join/#asterisk idc-dutch (n=idc-dutc@idc-vls.xs4all.nl) |
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| 15:52.04 | csiadmin | Corydon76-dig, I just going down this route as I've static/distortion on all pri calls |
| 15:52.28 | Corydon76-dig | csiadmin: please call Digium support |
| 15:52.30 | csiadmin | Corydon76-dig, and I've noticed the misses in dahdi_tool |
| 15:52.59 | csiadmin | Corydon76-dig, will do, thanks for your help. |
| 15:53.59 | *** join/#asterisk idc-dutch (n=idc-dutc@idc-vls.xs4all.nl) |
| 15:54.23 | idc-dutch | Anyone successful integrated asterisk 1.6.1 with OCS 2007 R2? |
| 15:55.02 | Corydon76-dig | idc-dutch: not until OCS supports UDP |
| 15:55.04 | feder | does anybody knows if is there a way to do inserts through realtime? |
| 15:55.04 | idc-dutch | eh? Asterisk 1.6 supports TCP? |
| 15:55.29 | lancey | idc-dutch: no |
| 15:55.33 | Corydon76-dig | idc-dutch: experimentally, yes |
| 15:55.51 | *** join/#asterisk Iamnacho (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 15:55.56 | Corydon76-dig | feder: yes, it's the ast_store_realtime() method |
| 15:56.01 | jaytee | I'm still using sipX as a proxy to Exchange 2007 UM until 1.6 TCP implementation is rock solid |
| 15:56.02 | [TK]D-Fender | idc-dutch: http://www.google.ca/search?hl=en&q=Asterisk+OCS+2007&btnG=Google+Search&meta=&aq=f&oq= <--------- |
| 15:56.43 | idc-dutch | [TK]D-Fender: Yes, got that one, but would like to discuss it with someone who actually did it |
| 15:56.46 | [TK]D-Fender | I love it when people ask questions that take less typing to get an immediate answer from Google for... |
| 15:57.24 | jaytee | I love it when a plan comes apart.....um, ooops, I meant together...yeah, that's what I meant |
| 15:57.34 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 15:57.34 | feder | Corydon76-dig: thx, but I need to do it from the dialplan |
| 15:57.43 | idc-dutch | jaytee: did you test with the TCP in 1.6? |
| 15:57.45 | Corydon76-dig | feder: No, sorry |
| 15:57.56 | rue_mohr | so the problems started with the fax machine not being able to dial out when I changed 4 of the sip accounts from info to rfc2833.... I wonder what really caused it... |
| 15:57.59 | Corydon76-dig | feder: Realtime is only really queryable from the dialplan |
| 15:58.04 | [TK]D-Fender | jaytee: I love it when a plan (not mine) comes apart... hence my love of Firefly |
| 15:58.21 | Corydon76-dig | feder: why don't you use func_odbc, if you need database integration? |
| 15:58.33 | [TK]D-Fender | feder: Realtime jsut sits in a DB and there are plenty of DB tools to let you do that. |
| 15:58.42 | [TK]D-Fender | feder: Corydon76-dig just handed you one |
| 15:58.43 | jaytee | idc-dutch, yes with 1.6.1.0 but it was problematic and I didn't have alot of time to dig into it in depth. |
| 15:58.53 | feder | Corydon76-dig: thx, I'll try that way |
| 15:59.34 | feder | Corydon76-dig: It feels weird to be able to do SELECT and UPDATES but no INSERTS |
| 15:59.37 | idc-dutch | jaytee: can you share what problems you ran into? I'm just running into 500 responses from the mediation server, is that what you ran into too? |
| 15:59.50 | jaytee | idc-dutch, there've been significant improvements and fixes since but from what others have mentioned in here there are still issues. |
| 16:00.27 | idc-dutch | any place I can see those issues, so I can compare what I run into? |
| 16:00.30 | jaytee | idc-dutch, my results would not pertain to what you're doing since I just use Exchange UM and don't use the mediation server or OCS itself |
| 16:00.44 | jaytee | idc-dutch, hang on a second |
| 16:01.11 | idc-dutch | thx. Just wondering if those issues are stability, or just preventing the whole thing all together |
| 16:02.02 | idc-dutch | Stability issues aren't a prob really, since it's a test enviroment, but currently I'm somehow stuck all together. And yes, I hate OCS too, but I need to set this up. |
| 16:05.00 | jaytee | check out this blog, it pertains to Mediation server with Asterisk and has other useful links. http://blog.tmcnet.com/blog/tom-keating/asterisk/asterisk-with-office-communications-server-2007.asp |
| 16:06.47 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 16:06.48 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 16:07.01 | jaytee | lunchtime!!! |
| 16:07.09 | idc-dutch | kk, read that one several times already, but thx ;) |
| 16:07.40 | idc-dutch | Must be the frikking OCS that is toying with me then I guess |
| 16:08.39 | rue_mohr | if I make a call between two dahdi channels, shouldn't echocancelwhenbridged=no make sure the hwec is OFF!? |
| 16:11.33 | rue_mohr | arg, they cant use the fax machine and I dont know whats causing the muting of the dialing |
| 16:11.39 | VoipForces | does not see the point in connecting asterisk with OCS and all that UC stuff. |
| 16:13.40 | *** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net) |
| 16:13.40 | *** mode/#asterisk [+o mog] by ChanServ |
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| 16:16.28 | VoipForces | idc-dutch: What will you gain by adding OCS? Curious... |
| 16:18.10 | [TK]D-Fender | ulcers :) |
| 16:19.04 | VoipForces | [TK]D-Fender: Yeah I know. I was asked to setup this in lab mut been putting it off cause I do not see the advantages... and I hate waistig my time |
| 16:25.11 | GrantA | is there a number for the UK that can read back your CID |
| 16:27.13 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 16:29.41 | rue_mohr | help I need the echo canceler OFF in dahdi->dahdi calls! |
| 16:30.22 | Aiatek | turn off in chan_dahdi.conf, by the way it comes off by default |
| 16:30.26 | [TK]D-Fender | rue_mohr: "echocancelwhenbridged=no" <- duh :p |
| 16:30.33 | rue_mohr | thats what I have! |
| 16:30.39 | rue_mohr | I said it twice! |
| 16:30.43 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 16:30.56 | [TK]D-Fender | rue_mohr: sshow me that its on. |
| 16:31.02 | rue_mohr | ugh |
| 16:31.07 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 16:32.32 | rue_mohr | http://www.pastebin.ca/1453875 <- now do you belive me? |
| 16:32.42 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 16:34.38 | rue_mohr | wait, why is nativebridge=0? |
| 16:34.38 | [TK]D-Fender | rue_mohr: Show me that EC is actually IN USE :) |
| 16:34.48 | rue_mohr | how the hell do you expect me to do that!? |
| 16:34.50 | [TK]D-Fender | rue_mohr: and yeah... that looks SMRT too :) |
| 16:35.06 | [TK]D-Fender | rue_mohr: "dahdi show channel X" |
| 16:35.06 | rue_mohr | you might as well ask me to prove the moons not made of cheese |
| 16:35.22 | [TK]D-Fender | rue_mohr: You don't even know how to look at a channel.. sheesh :p |
| 16:35.29 | rue_mohr | cat /sys/module/wctdm24xxp/parameters/nativebridge |
| 16:35.30 | rue_mohr | 0 |
| 16:35.33 | rue_mohr | ? |
| 16:35.49 | rue_mohr | dosn't that mean asterisk is shoveling the data between dahdi channels? |
| 16:35.55 | rue_mohr | manually |
| 16:37.35 | [TK]D-Fender | rue_mohr: Look at a channel while bridged |
| 16:37.58 | rue_mohr | I'm trying to figure out how to make a bridged call I can still reach the computer in time to catch |
| 16:38.49 | rue_mohr | aha! |
| 16:38.53 | rue_mohr | it IS on! |
| 16:39.41 | rue_mohr | http://www.pastebin.ca/1453894 |
| 16:40.22 | [TK]D-Fender | rue_mohr: Show me the channels |
| 16:40.35 | [TK]D-Fender | rue_mohr: "core show channels concise" <- |
| 16:41.18 | rue_mohr | debian*CLI> core show channels concise |
| 16:41.18 | rue_mohr | debian*CLI> |
| 16:41.33 | rue_mohr | oh while in use still |
| 16:41.37 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 16:41.56 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 16:41.58 | *** join/#asterisk mohawk (n=ross@host217-40-110-153.in-addr.btopenworld.com) |
| 16:42.17 | rue_mohr | it hung up on me |
| 16:42.20 | rue_mohr | geeesh |
| 16:43.03 | rue_mohr | http://www.pastebin.ca/1453900 |
| 16:43.37 | rue_mohr | cordless to 26, which is the fax line |
| 16:43.48 | [TK]D-Fender | rue_mohr: try again and actually RREDUMP the channels... |
| 16:43.50 | [TK]D-Fender | asjashgdagslfd |
| 16:46.04 | rue_mohr | http://www.pastebin.ca/1453904 |
| 16:46.14 | rue_mohr | see! echo is on! |
| 16:47.09 | rue_mohr | and its a problem cause its brief muting is screwing up the dialing |
| 16:47.45 | *** join/#asterisk s14ck (n=s14ck@190.38.214.204) |
| 16:47.57 | rue_mohr | Echo Cancellation: 128 taps unless TDM bridged, currently ON <- aka, its not bridged, why. |
| 16:48.28 | rue_mohr | cat /sys/module/wctdm24xxp/parameters/nativebridge |
| 16:48.28 | rue_mohr | 0 |
| 16:48.34 | rue_mohr | I ahve my suspicions |
| 16:49.11 | rue_mohr | echo's 1 to it |
| 16:49.48 | *** join/#asterisk lucasb (n=bussey@office.telifon.com) |
| 16:50.23 | rue_mohr | no diff |
| 16:50.36 | rue_mohr | why cant this do local bridges |
| 16:50.47 | rue_mohr | can a tdm800 do local bridges? |
| 16:51.39 | rue_mohr | this is frustrating, i have newer dahdi drivers I cant install cause everyone is on the phone |
| 16:54.39 | rue_mohr | so your stumped |
| 16:55.06 | *** join/#asterisk s14ck (n=s14ck@ccscliente156.ifxnetworks.net.ve) |
| 16:56.14 | rue_mohr | hey does ringdebounce cause a delay between the fxo line ringing and the start of processsing it!? |
| 16:57.23 | russellb | that's what debouncing is by definition :-) |
| 16:58.20 | rue_mohr | ok just to confirm ringdebounce is called debouncetime? |
| 16:58.57 | rue_mohr | 1024! |
| 16:59.07 | rue_mohr | nowonder there is such a delay |
| 16:59.11 | russellb | that is most likely samples. |
| 16:59.22 | russellb | not ms |
| 16:59.26 | rue_mohr | its ms in the config |
| 16:59.37 | russellb | shrugs |
| 16:59.38 | rue_mohr | 1024 is the /sys/module/wctdm24xxp/parameters/ringdebounce |
| 16:59.48 | russellb | feel free to contact digium support if you'd like any further clarification |
| 16:59.55 | rue_mohr | :) |
| 17:00.08 | rue_mohr | I'm trying not to bother them too much without having had training |
| 17:01.55 | Qwell | http://www.digium.com/en/training/ :D |
| 17:02.12 | rue_mohr | I know I know |
| 17:02.24 | rue_mohr | look at it this way, I'm learning |
| 17:02.38 | rue_mohr | I consider myself quite officially dangerous |
| 17:02.57 | Aiatek | <@Qwell> who takes the trainning has any priviliges in digium? |
| 17:03.09 | Qwell | huh? |
| 17:03.14 | Aiatek | :P |
| 17:03.43 | Aiatek | like 25% of the company or something |
| 17:03.45 | Qwell | Aiatek: I hear people that take training courses can get their book signed by Jared Smith |
| 17:03.46 | Qwell | ~book |
| 17:03.47 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 17:03.54 | Aiatek | hahahahahhahahhaha |
| 17:04.06 | rue_mohr | I'd be really greatfull if somone has ideas why a) asterisk isn't bridging dahdi->dahdi calls, and why its not turning off the echo canceler for those calls |
| 17:04.16 | rue_mohr | I ahve the book, it dons't cover this |
| 17:04.24 | Qwell | echocancelwhenbridged |
| 17:04.35 | rue_mohr | I'll show you my config |
| 17:04.42 | rue_mohr | http://www.pastebin.ca/1453875 |
| 17:04.45 | rue_mohr | its off |
| 17:04.56 | rue_mohr | http://www.pastebin.ca/1453904 |
| 17:05.09 | rue_mohr | as you can see from there, its on when the call should be bridged |
| 17:05.11 | Qwell | so...then...why would it try to cancel echo on bridged calls if you've disabled it? |
| 17:05.31 | rue_mohr | cause its not off |
| 17:05.48 | *** join/#asterisk metfan2007 (n=jc@201.103.101.11) |
| 17:05.59 | rue_mohr | and its muting the dtmf of hte fax machine, and the digits dont go to the co, and the boss wants to send his fax |
| 17:06.14 | rue_mohr | and is rather frustrated that he cant right now |
| 17:06.27 | ariel_ | fax setups should have no EC |
| 17:06.32 | metfan2007 | hi all, I have some problems with queues, I'm getting some NOTE IN USER agent status, and they cannot receive calls, what does that means? |
| 17:06.33 | rue_mohr | REALLY!?!?! |
| 17:06.39 | metfan2007 | NOTE IN USE |
| 17:06.48 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 17:07.01 | rue_mohr | metfan2007, whats your limit set to? |
| 17:07.04 | [TK]D-Fender | rue_mohr: How is it that is should mute DTMF? When your device dials, it sin't bridged because it hasn't even hit the dialplan yet <- |
| 17:07.41 | rue_mohr | the echo cancel does a brief mute when it hears audio, the fax machine only send a brief dtmf tone, they dont work togethor |
| 17:07.49 | rue_mohr | there is no dialplan |
| 17:07.53 | metfan2007 | rue_mohr, 1 |
| 17:08.13 | rue_mohr | when the fax machine picks up, its automatically connected to dahdi/6 |
| 17:08.34 | rue_mohr | and its digits go straight to the co |
| 17:08.47 | [TK]D-Fender | rue_mohr: lovely. |
| 17:08.57 | metfan2007 | rue_mohr, they are not in call, and I have 10 calls of clients in que queue waiting for an avaiable agent |
| 17:09.04 | rue_mohr | where the echo can effectivly mutes out all the digits |
| 17:09.13 | [TK]D-Fender | metfan2007: And you should be pastebining everything.... |
| 17:09.21 | rue_mohr | metfan2007, whats the limit set to? |
| 17:11.30 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 17:12.16 | rue_mohr | ok, so everyone agrees that the fax should have no ec, that echocancelwhenbridged is off, and that the echo canceler is on on a cal between dahdi channels |
| 17:13.01 | rue_mohr | so nobody has a fix |
| 17:13.02 | [TK]D-Fender | rue_mohr: Even if EC is enabled, DAHDI should disable it upon hearing a fax tone |
| 17:13.13 | rue_mohr | ah |
| 17:13.20 | rue_mohr | I was testing with a cordless set |
| 17:13.24 | [TK]D-Fender | rue_mohr: taht is automatic and can't be overridden |
| 17:13.28 | rue_mohr | BUT |
| 17:13.42 | rue_mohr | I need the ec off before that because its the dialing thats being screwed up |
| 17:14.04 | rue_mohr | the ec is destroying hte fax machines dialing dtmf |
| 17:14.18 | ariel_ | rue_mohr: also immediate=yes, it's normally used for like door phones and overhead paging system. (at least in all my setups) |
| 17:14.34 | rue_mohr | ok, is echocancel=yes a per channel setting? |
| 17:15.29 | [TK]D-Fender | rue_mohr: pastebin your configs again |
| 17:15.40 | rue_mohr | http://www.pastebin.ca/1453875 |
| 17:15.47 | [TK]D-Fender | ariel_: He's passing raw tone. |
| 17:15.58 | [TK]D-Fender | rue_mohr: Why is this even going through * ? |
| 17:16.10 | rue_mohr | cause the sip phones use the line as an extra |
| 17:16.10 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 17:16.34 | rue_mohr | and its going to be gaining distinctive ring as anohter business line |
| 17:16.41 | [TK]D-Fender | rue_mohr: Ok, you've flat-lines the gains again? |
| 17:16.48 | ariel_ | argh |
| 17:17.01 | rue_mohr | I flatlined the fxs gains |
| 17:17.11 | rue_mohr | not the fxo gains |
| 17:17.12 | [TK]D-Fender | rue_mohr: All from what I have seen |
| 17:17.24 | [TK]D-Fender | ah, jnoe |
| 17:17.27 | [TK]D-Fender | nope* |
| 17:17.34 | rue_mohr | first two channel defs are fxs |
| 17:17.39 | [TK]D-Fender | rue_mohr: As a sanity check, go try that |
| 17:17.52 | rue_mohr | try which? |
| 17:18.00 | rue_mohr | I'm using that now, no diff |
| 17:18.20 | rue_mohr | I'm gonna try echocancel=off also for the first two |
| 17:18.32 | [TK]D-Fender | rue_mohr: flat-line them all. |
| 17:18.32 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 17:18.47 | rue_mohr | I cant, the office people wont be abel to hear the callers |
| 17:19.19 | spck | anyone got heartbeat and drbd setup with their * box? |
| 17:19.23 | rue_mohr | I have them dialed like that for a reason! |
| 17:19.37 | rue_mohr | spck, I was asking abotu that a while ago |
| 17:19.48 | rue_mohr | your interested it failsafe? |
| 17:20.08 | rue_mohr | would a pulse train out a serial or parallel port work for you? |
| 17:20.31 | rue_mohr | what kinda failure notification delay can you work with? |
| 17:20.32 | spck | <PROTECTED> |
| 17:20.47 | spck | just the nic |
| 17:20.51 | rue_mohr | by what mechanism... ah |
| 17:21.10 | rue_mohr | you might be able to poll a sip port |
| 17:21.38 | spck | i got this problem where i shutdown nodeA and nodeB takes over like it should, but then when i bring nodeA back up it takes over for a second and then goes passive again |
| 17:21.43 | metfan2007 | rue_mohr, [TK]D-Fender http://pastebin.ca/1453970 |
| 17:21.45 | rue_mohr | I'm looking for a hardware signal for my system at home |
| 17:22.30 | rue_mohr | so your limit is 1 |
| 17:22.30 | spck | it's like the opposite of split brain |
| 17:22.44 | rue_mohr | I cant help, i'm a noob |
| 17:23.07 | rue_mohr | I can sometimes point properly though |
| 17:23.15 | metfan2007 | [TK]D-Fender http://pastebin.ca/1453970 |
| 17:23.35 | [TK]D-Fender | metfan2007: Doesn't show me the problem. |
| 17:24.01 | rue_mohr | with no calls, his operators are busy |
| 17:24.14 | rue_mohr | get his to show you his channels? |
| 17:25.03 | metfan2007 | [TK]D-Fender: Take a look to this http://pastebin.ca/1453980 |
| 17:25.16 | rue_mohr | arg, every time I have an opportunity to fix somehting, everyone picks up the phone |
| 17:26.16 | metfan2007 | [TK]D-Fender: 10 callers waiting for an available agent, and 5 agents not in call, waiting for a caller, but in "Not in use" state... :( |
| 17:26.54 | rue_mohr | so, are you guys saying I should have asterisk collect the digits and then send them to the co, even though I need NO translation? |
| 17:27.18 | rue_mohr | metfan2007, sip show channels, see if their phones show up when their not busy |
| 17:28.15 | [TK]D-Fender | metfan2007: You have no timeouts configured, etc... |
| 17:29.00 | [TK]D-Fender | metfan2007: No inter-agent delay, no wrapup.... go configure these basics as per the sample config |
| 17:29.22 | metfan2007 | [TK]D-Fender ok ok |
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| 17:43.16 | rue_mohr | ok, hah, I was blaming the wrong thing, cause echo cancel is off and its still brifly muting the dtmf |
| 17:44.06 | rue_mohr | can anyone help me? |
| 17:44.06 | rue_mohr | this really isn't something I can do over the phone with digium |
| 17:45.17 | rue_mohr | is echocancel in chan_dahdi.conf a per line setting! |
| 17:45.23 | rue_mohr | ??? |
| 17:45.36 | spck | shrugs |
| 17:45.45 | rue_mohr | I dont knwo which of these settings are set-once and which are per cahnnel |
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| 17:47.36 | bmoraca | rue_mohr: in your dial string, are you using any options (t, T, etc)? if so, i believe it won't bridge, just like SIP calls won't reinvite |
| 17:47.45 | *** part/#asterisk lanning (n=lanning@173.8.187.197) |
| 17:47.50 | neurosys | I got my hands on a Linksys SPA942 to play with. Incomming from ITSP thru My asterisk works just fine. But when i place calls from the phone, There is no audio on either end.. UNLESS i place the call on hold. Then the outbound call hears the hold music and once released, we both hear each other! Ive been googling and twisting the configs to no avail. Any suggestions? |
| 17:49.00 | rue_mohr | oh? |
| 17:49.06 | rue_mohr | no it just dials |
| 17:49.13 | rue_mohr | just a sc |
| 17:49.31 | rue_mohr | neurosys, check your rtp |
| 17:50.20 | rue_mohr | http://www.pastebin.ca/1454031 |
| 17:50.33 | rue_mohr | so your saying that it should have an option on there eh? |
| 17:50.40 | bmoraca | no |
| 17:50.46 | rue_mohr | oh |
| 17:50.55 | bmoraca | i'm saying that if it DOES have an option, the call HAS to go through Asterisk |
| 17:51.25 | bmoraca | rue_mohr: show me the context your cordless phone is in |
| 17:51.34 | rue_mohr | what about if I globally have a feature turned on |
| 17:51.39 | rue_mohr | k |
| 17:53.14 | rue_mohr | bmoraca, http://www.pastebin.ca/1454036 |
| 17:53.59 | rue_mohr | exten 26 is your focus point |
| 17:54.22 | rue_mohr | lFAX=DAHDI/6 |
| 17:54.53 | bmoraca | weird |
| 17:55.30 | rue_mohr | I have the echo canceler off now, I have echocancel=off for hte fxs channels |
| 17:55.45 | bmoraca | generally a good idea for fax channels |
| 17:56.00 | leifmadsen | not generally -- is. |
| 17:56.44 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 17:57.14 | rue_mohr | and calls between a sip and fxs still seem ok |
| 17:57.29 | rue_mohr | aside from an overabundance of side tone on the sip side |
| 17:58.29 | bmoraca | shit...vmware converter doesn't work with NT4 |
| 17:59.00 | *** part/#asterisk Fabian- (n=Fabian@speeldoos.duocast.nl) |
| 17:59.15 | rue_mohr | could the dynamic feature I have set be causing this? |
| 17:59.35 | rue_mohr | can I disable a dynamic feature without completely restarting asterisk? |
| 18:00.30 | rue_mohr | are dynamic features limited to contexts? |
| 18:02.28 | [TK]D-Fender | rue_mohr: No, they are limited by what you set in the channel var |
| 18:02.51 | [TK]D-Fender | neurosys: Sounds like a classic NAT/reinvite issue |
| 18:10.31 | nkohh | wishes he had a ClueBat (tm) |
| 18:14.47 | [TK]D-Fender | lets him have it.... |
| 18:15.00 | [TK]D-Fender | nkohh: WHAM! wham! WHAM! wham! WHAM! wham! WHAM! wham! WHAM! wham! WHAM! wham! WHAM! wham! WHAM! wham! |
| 18:15.31 | *** join/#asterisk stijnbe (n=stijnbe@78-21-61-204.access.telenet.be) |
| 18:18.20 | neurosys | [TK]D-Fender: Hmmm. Is this a bug or do i need to use the canreinvite switch in sip.conf/ |
| 18:19.10 | [TK]D-Fender | ~sipnat |
| 18:19.10 | infobot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 18:19.11 | [TK]D-Fender | ^^^^^^ |
| 18:19.14 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 18:19.23 | neurosys | thx [TK] |
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| 18:34.08 | jaytee | I kind of imagine that if BamBam Rubble had gone into IT when he grew up then he'd be just like [TK]D-Fender |
| 18:34.26 | *** part/#asterisk korihor (n=korihor@190.72.254.245) |
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| 18:35.57 | [TK]D-Fender | BAM BAM! |
| 18:36.20 | nkohh | i dont think so |
| 18:36.26 | nkohh | i get the distinct impression that d-fender is canadian |
| 18:36.42 | jaytee | Pebbles on the other hand I always imagined sliding around on a pole in a "gentleman's club" |
| 18:38.04 | jaytee | nkohh, he's French Canadian, it's like a regular Canadian but you add alot more butter and cheese |
| 18:38.51 | [TK]D-Fender | jaytee: I'm an English Canadian... I just happen to speak french :) |
| 18:38.55 | nkohh | ROFL. |
| 18:39.17 | [TK]D-Fender | nkohh: want some more impressions? ;) |
| 18:39.27 | nkohh | [TK]D-Fender: eh? |
| 18:39.30 | nkohh | just kidding |
| 18:39.37 | nkohh | is canadian and can smell fellow canadians from miles away |
| 18:39.41 | [TK]D-Fender | reaches for his ClueBat (tm) |
| 18:39.50 | jaytee | Lobster in New England used to be considered a poor person's food and beneath the privelidged class until the French showed us how to prepare it properly with clarified butter. |
| 18:41.08 | spck | only the french could make eating giant cockroaches fashionable |
| 18:41.37 | ariel_ | Hummm, Lobster good...., any way you cook them. |
| 18:41.48 | jaytee | mmmmm, you betcha! |
| 18:41.58 | nkohh | doesn't like lobster or crab. |
| 18:42.08 | ariel_ | good more for me. |
| 18:42.16 | nkohh | ;) |
| 18:43.24 | jaytee | travel section of MSNBC.com had an article about the 10 best lobster places in Maine. I've eaten at 4 of them. Being stuck out here in Indianapolis so far from home the article made me want to cry. |
| 18:43.30 | beek | jaytee: Enough of the seafood talk, dammit! You're making me hungry. |
| 18:43.46 | beek | I'm in the middle of PA so the best I can do for seafood is "Red Lobster" -- 'nuff said. |
| 18:44.17 | jaytee | ok, yeah that would suck |
| 18:44.19 | ariel_ | lol, I am in the south and we eat allot of fake lobsters called longustino's |
| 18:44.50 | jaytee | and those dick tease excuses for lobster, crawdads. tasty but it takes you hours just to fill up on em |
| 18:45.02 | jaytee | but I'll shut up about food now :-) |
| 18:46.31 | beek | tries to concentrate on this programming issue while dreaming of lobster... he thanks jaytee. |
| 18:47.38 | jaytee | aww, man! I apologize. but how do you think I feel so far from New England. All they eat around here is deep fried pig nuts and bratwurst |
| 18:47.58 | nkohh | is going to get some aloo gobi and muttar paneer for lunch |
| 18:47.59 | nkohh | think about that boys |
| 18:48.01 | nkohh | think about that. |
| 18:48.17 | jaytee | I have no idea what either of those dishes are and I'm not sure I want to |
| 18:48.29 | nkohh | oh you do |
| 18:48.36 | nkohh | east indian cuisine is the best cuisine |
| 18:48.58 | beek | Didn't Carlin do a bit about food that sounds like what it is? If so, I hate to ask... |
| 18:49.20 | jaytee | lemme guess, the brains of some small primate served either raw with a sprig of parsely or chilled with a side of rice |
| 18:49.43 | beek | jaytee: You should be able to get the key ingredients of that dish. |
| 18:50.32 | jaytee | beek, well yeah if you like lemur brains or gibbon brains but then lemurs aren't really primates I don't think |
| 18:50.52 | nkohh | jaytee: http://en.wikipedia.org/wiki/Mattar_paneer http://en.wikipedia.org/wiki/Aloo_gobi |
| 18:51.11 | beek | Of course, there's an elementary school about a 1/2 mile from here so small primate brains wouldn't be that difficult to obtain. |
| 18:51.26 | ariel_ | is now just thinking of a nice BabyBack Ribs to get all those other foods out of his mind.... |
| 18:51.42 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 18:52.09 | beek | nkohh: they both sound tasty. |
| 18:54.33 | *** join/#asterisk aurax (n=aurax@bzq-84-108-166-31.cablep.bezeqint.net) |
| 18:54.38 | aurax | hello mates |
| 18:55.05 | *** join/#asterisk Errotan (n=Errotan@5403E4F1.catv.pool.telekom.hu) |
| 18:55.35 | aurax | can anyone assist, i'm looking for a voip (sip) provider that will supply me with local phone lines in other countries? beside net2phone - they suck. |
| 18:56.07 | jaytee | ok, both those dishes sound delicious, nothing weird in them at all |
| 18:56.16 | nkohh | aurax: many providers will do that for you. |
| 18:56.25 | jaytee | the Mattar paneer sounds very tasty |
| 18:56.30 | nkohh | aurax: there is a nice list here: http://www.voip-info.org/wiki/view/DID+Service+Providers |
| 18:56.35 | jaytee | I like vindaloo |
| 18:57.17 | *** join/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 18:57.43 | *** join/#asterisk bakermd (n=bakermd@38.101.225.215) |
| 18:57.59 | bakermd | Where can I get the latest firmware for my Polycom phones? |
| 18:58.11 | ariel_ | polycom |
| 18:58.26 | bakermd | There is a CentOS Repo for Trixbox users that has the 3.0 fw in it - I would prefer the 4.0 though |
| 19:00.21 | bakermd | The Release Matrix on Polycom's site does not list 4.x firmware |
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| 19:02.18 | bakermd | nm - was not reading enough |
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| 19:06.57 | *** mode/#asterisk [+o denon] by ChanServ |
| 19:10.46 | *** join/#asterisk intralanman (n=lanman@static-76-160-148-146.dsl.cavtel.net) |
| 19:11.11 | [TK]D-Fender | jaytee: Not sure I've had Vindaloo before... I've seen it on a menu listed as "may cause not-so-spontaneous combustion" |
| 19:12.13 | [TK]D-Fender | nkohh: And this doubles as #cuisine , for which there are many connoisseurs here |
| 19:13.04 | nkohh | apologizes for dragging cuisine into #asterisk |
| 19:14.36 | [TK]D-Fender | nkohh: Was here long before you... |
| 19:16.02 | *** join/#asterisk CrazyTux (n=brandon@ip98-164-236-47.oc.oc.cox.net) |
| 19:16.37 | CrazyTux | Hey guys -- anyone ever experience an issue i.e. with a PAP2 / VoIP Analog -> VOIP Adapter, where the digits pressed on the phone are truncated / or replicated on what the actual VoIP Device Receives. |
| 19:17.02 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:17.12 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-37500cefae2044b1) |
| 19:17.30 | nkohh | [TK]D-Fender: I mean this particular instance :) |
| 19:17.31 | ariel_ | dtmf settings can do that as well as nat issues, and yes there are more things that can do this to them. |
| 19:18.51 | *** join/#asterisk jicksta (n=jicksta@67.164.0.78) |
| 19:19.17 | jaytee | nkohh, nope! it's my fault, not yours. I was the one that started running off about crustaceans |
| 19:20.36 | Deeewayne | seanbright, Qwell: thanks! |
| 19:20.55 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 19:21.06 | seanbright | eh? |
| 19:21.31 | Deeewayne | conversation you two had last august about: |
| 19:21.37 | Deeewayne | ~centos52bug |
| 19:21.37 | infobot | [~centos52bug] There is a bug compiling Zaptel up to 1.4.11/1.2.26 on CentOS/RHEL 5.2: error in xpp/xdefs.h. Workaround: uncheck "xpp" (Astribank) driver in menuselect. Fixed in 1.4.12. Trouble compiling x86_64 packages? Report a bug to CentOS! ".i386 packages should not satisfy dependencies for .x86_64 packages." |
| 19:21.37 | nkohh | jaytee: just as well ;) |
| 19:22.03 | Deeewayne | I'm suffering from it ^ now |
| 19:22.08 | seanbright | ah. yeah. you're welcome :) |
| 19:22.14 | seanbright | yum remove *.i366 |
| 19:22.19 | seanbright | yum remove *.i386 |
| 19:22.21 | seanbright | the latter |
| 19:27.40 | CrazyTux | ariel_: it wouldnt be NAT I'm talking merely from the Phone / ATA |
| 19:27.44 | joako | Anyone use XO Flex PRI? I want to bypass the PRI and use MGCP directly... |
| 19:28.23 | *** join/#asterisk smps (n=smps@193.170.53.51) |
| 19:32.13 | *** join/#asterisk xfighter (n=xfighter@a95-198.adsl.paltel.net) |
| 19:32.27 | xfighter | good evening ladies and gentlemen |
| 19:33.24 | xfighter | I have a question if you may |
| 19:34.38 | xfighter | anybody here to respond? |
| 19:36.08 | kaldemar | ~ask |
| 19:36.08 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 19:36.17 | *** join/#asterisk dacs (n=dacs@unaffiliated/dacs) |
| 19:36.17 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 19:36.17 | smps | xfighter, just ask, if someone knows it he/she will answer etc .... |
| 19:36.18 | xfighter | hello kaldemar |
| 19:36.34 | xfighter | thank you all |
| 19:36.41 | xfighter | it's very simple |
| 19:36.42 | dacs | i am trying to run asterisk -vvvc and i get Segmentation fault |
| 19:36.58 | seanbright | dacs: that is by design. |
| 19:37.15 | xfighter | you see I applied all the asterisk book instructions |
| 19:37.21 | xfighter | for ch3 |
| 19:37.28 | xfighter | installin the system |
| 19:37.36 | xfighter | I just wanna know how can I make sure that everything I done correctly untill now?? |
| 19:38.14 | seanbright | dacs: what version of asterisk? |
| 19:38.17 | dacs | seanbright: Unable to connect to remote asterisk (does /var/run/asterisk.ctl |
| 19:38.45 | dacs | seanbright: Asterisk 1.6.0.6 |
| 19:39.02 | seanbright | dacs: step 1, upgrade to 1.6.0.10 |
| 19:39.03 | *** join/#asterisk [acer]lanman (n=lanman@68-242-127-131.pools.spcsdns.net) |
| 19:39.41 | *** join/#asterisk AD2008 (n=RatX@wbs-41-208-238-225.wbs.co.za) |
| 19:39.52 | dacs | seanbright: i just got this to work with my Debian |
| 19:39.53 | dacs | lol |
| 19:39.54 | ariel_ | xfighter: start using it and see how it works |
| 19:40.08 | seanbright | dacs: yeah, sounds like it is working perfectly |
| 19:40.14 | seanbright | with the whole "segmentation fault" thing |
| 19:40.18 | ariel_ | dacs: when you do asterisk -vvcg it will tell you what module it crashed on |
| 19:40.25 | xfighter | but ariel it's not complete yet |
| 19:40.28 | dacs | seanbright: hahahhaahah |
| 19:40.28 | xfighter | I can't test it |
| 19:41.06 | dacs | a bunsh of them |
| 19:41.09 | ariel_ | xfighter: then you have more rules and setup to do |
| 19:41.19 | ariel_ | xfighter: I would start simple then work from there |
| 19:41.26 | seanbright | dacs: can you pastebin the output of running 'asterisk -cvvvvvvvvvg' |
| 19:41.28 | seanbright | ~pb |
| 19:41.28 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 19:41.48 | ariel_ | dacs: last few are what are causing your issue. |
| 19:42.10 | seanbright | dacs: ariel_ will help you from here on out |
| 19:42.39 | ariel_ | seanbright: really? |
| 19:43.04 | seanbright | you won't? |
| 19:43.17 | seanbright | huh, strange. |
| 19:43.34 | ariel_ | like allot of us I am doing 3 or 4 things at the same time so if I am looking at this screen I can help. |
| 19:43.51 | ariel_ | but yes I will help as much as I can |
| 19:44.07 | seanbright | well he is all yours |
| 19:44.09 | dacs | seanbright: http://pastebin.com/d76ac4e44 |
| 19:44.10 | seanbright | good luck and god speed. |
| 19:45.34 | dacs | seanbright: why? :( |
| 19:46.04 | xfighter | ariel |
| 19:46.10 | xfighter | one more question please |
| 19:46.29 | xfighter | if am not planning to connect my * to the landline telephone network |
| 19:46.38 | xfighter | do I need to configure zaptel |
| 19:46.44 | dacs | ariel_: http://pastebin.com/d76ac4e44 this is the output of asterisk -vvvvvcg |
| 19:46.54 | xfighter | or I just need to configure SIP only? |
| 19:47.11 | cvnet | does anyone here uses h323 on their * box? Im having problem connecting to Virizon, I was wondering if I can send a test call to anyone to test h323 |
| 19:47.57 | ariel_ | dacs: is this a new setup? |
| 19:48.04 | dacs | ariel_: yes |
| 19:48.17 | AD2008 | cvnet - no, but are you getting 1 way audio or no audio? check for UDP port range mismatch |
| 19:48.28 | ariel_ | xfighter: it's up to you, But I would start basic and work from there. |
| 19:48.51 | xfighter | what do you mean by start basic and work from there? |
| 19:48.52 | ariel_ | dacs: do you have the actual core dump image? |
| 19:49.18 | dacs | ariel_: how can i check |
| 19:49.21 | ariel_ | xfighter: if you set the system up basic and do make samples it's really almost ready to test. |
| 19:50.16 | *** join/#asterisk stope (n=nobody@sud-cable-cmts3-69-60-242-213.vianet.ca) |
| 19:50.19 | cvnet | AD2008: I only get busy signal, when i contacted Virizon I was told -->Your calls are showing multiple timers that are incompatible. (For example packetization timer of 150 ms on g711Ulaw64k and g711Alaw64k calls). These will be rejected by our ENGINE platform. You are sending in incompatible packetization timing on your OLC (open logical channel)'s codec list. Our server is replying with a "normal unspecified" RELEASE COMPLETE message within millisecond |
| 19:50.36 | kaldemar | xfighter: there are some things such as MeetMe (a conference application) and IAX2 trunking (IAX2 VoIP protocol trunking, i.e. reducing overhead) that require timing from a zaptel module (zaptel has been renamed to DAHDi btw.). but you won't need those with basic calls with SIP and you can always recompile with zaptel/DAHDi later. |
| 19:50.46 | stope | anyone know why a linksys 2102 or spa942 would return a 486 busy status even though do not disturb is not activated? |
| 19:51.01 | ariel_ | dacs: this was the last thing to load, res_clioriginate.so => (Call origination from the CLI), did you setup your zaptel? or dahdi? |
| 19:51.19 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 19:51.19 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 19:51.36 | dacs | ariel_: none , i was following this http://mikeoverip.wordpress.com/2009/03/11/asterisk-16-compilation-and-installation-on-debian-5-lenny/ |
| 19:51.41 | AD2008 | cvnet - from the little i know, the timer should be either 120 or 150, should be a setting in a conf file somewhere you an set |
| 19:52.48 | AD2008 | h323.conf i would guess |
| 19:53.00 | cvnet | AD2008: G.711 Packetization 20ms <-- Virizon wants, i did some research on the net and found that you can change it on h323.conf like this allow=711:20 <-- but still nothing |
| 19:53.07 | [TK]D-Fender | xfighter: No you do not need zaptel, and you should ahve sip.cfon and extensions.conf configured to expect to be able to do anything |
| 19:54.08 | dacs | ariel_: are you able to help me ? |
| 19:54.09 | AD2008 | cvnet - maybe you need to do it in your h.323 channel driver? |
| 19:54.37 | cvnet | AD2008: hum, would i just open the file? how? |
| 19:55.08 | AD2008 | guess it would depend upon which one you are using, do you know? |
| 19:55.19 | AD2008 | i think openh323 is quite popular |
| 19:56.48 | cvnet | wheni make the calls i just use like exten => _X.,n,Dial(H323/16137451576@65.203.146.114) i guess im using h323 |
| 19:57.55 | ariel_ | dacs: without the core dump it's a hit or miss, the setup you posted is very basic and if all the depency's were installed it should work. Did you make samples? |
| 19:58.07 | cvnet | chan_h323.so is compiled cant be edited |
| 19:58.31 | spck | what's everyone's favorite way to provision polycom phones? |
| 19:58.53 | [TK]D-Fender | spck: vi <- |
| 19:58.58 | [TK]D-Fender | spck: + FTP |
| 19:59.33 | spck | that doesn't sound very easy |
| 19:59.38 | kaldemar | also uses dhcp server to pass boot server address |
| 19:59.57 | dacs | when i start * using asterisk -r i get Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) |
| 20:00.12 | dacs | and it does exist |
| 20:00.13 | Aiatek | try just asterisk |
| 20:00.17 | Aiatek | asterisk |
| 20:00.24 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-181-134.telkomadsl.co.za) |
| 20:00.24 | Aiatek | later asterisk -r |
| 20:00.47 | SuPrSluG | dacs: is res_clioriginte.so in /usr/lib/asterisk/modules/ |
| 20:00.58 | xfighter | [TK]D-Fender : WELCOME PROFESSOR |
| 20:01.23 | [TK]D-Fender | xfighter: And as I said I don't feel * is a good basis for your subject matter |
| 20:01.24 | dacs | SuPrSluG: yes |
| 20:01.45 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 20:01.54 | xfighter | I'd like to expand professor , but I ain't got lot of time to do so |
| 20:01.55 | SuPrSluG | dacs: are you upgrading? |
| 20:01.56 | [TK]D-Fender | xfighter: * is not a "voip server" per se |
| 20:02.01 | ariel_ | dacs: until asterisk starts correctly via the asterisk -vvvgc it's not going to allow you to do asterisk -r which is a remote connection |
| 20:02.10 | xfighter | unless you accept to guide me through it |
| 20:02.51 | xfighter | would you accept to be my guide?? |
| 20:03.31 | xfighter | professor? |
| 20:04.14 | ariel_ | xfighter: you might be better off doing pbx in a flash or asteriskNow iso which are all in one installs. |
| 20:05.15 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 20:05.44 | xfighter | ariel :D I don't like spoon feeding unless my time doesn't allow me I did download that iso but I guess the professor has good point of view that would be better for me unless it'll consumer more than a month of my time in this case I'll not be able to do H232 theory |
| 20:06.09 | [TK]D-Fender | xfighter: You are planning on giving a SEMINAR about VoIP tech and don't have the first clue about any given solution and no motivation. Sorry, but i have no intention of wasting my time doing your complete job for you :) |
| 20:06.23 | [TK]D-Fender | ~wglwat |
| 20:06.24 | infobot | from memory, wglwat is well, good luck with all that |
| 20:06.40 | xfighter | I didn't mean that |
| 20:06.52 | *** join/#asterisk mgroman (n=miles@unaffiliated/mgroman) |
| 20:06.54 | [TK]D-Fender | [16:02]<xfighter>unless you accept to guide me through it |
| 20:06.57 | cvnet | anyone here who uses h3232? I need to make a test call to see if my h232 is working or not |
| 20:07.01 | [TK]D-Fender | xfighter: riiiiiiiight |
| 20:07.03 | xfighter | but knowin nthn is not a shameful thing but learning it is a good thing |
| 20:07.08 | xfighter | yes I know nthn |
| 20:07.21 | xfighter | also yes I can learn |
| 20:07.22 | [TK]D-Fender | xfighter: And how soon do you need to be ready to give this seminar? |
| 20:07.33 | xfighter | also yes I need an experienced help |
| 20:07.46 | xfighter | not in everything but a nudge outta the door is helpful |
| 20:07.53 | SuPrSluG | @100/hr |
| 20:07.54 | xfighter | I have one year |
| 20:08.00 | xfighter | next spring |
| 20:08.04 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 20:08.11 | mgroman | Hi, my IVR system seems to stop working (phone just keeps ringing, never picks up) every couple of weeks or so. Usually a reboot of the server does the trick (which I know isnt right). Asterisk version 1.4.17 and Zaptel svn revision 4581, any ideas? |
| 20:08.12 | [TK]D-Fender | xfighter: That can be arranged :) |
| 20:08.16 | Deeewayne | O.O |
| 20:08.23 | mgroman | Or, How can I set up Asterisk to log errors ? |
| 20:08.30 | Deeewayne | prepares to watch the nudge |
| 20:08.36 | SuPrSluG | logger.conf |
| 20:08.40 | [TK]D-Fender | Deeewayne: :D |
| 20:08.40 | mgroman | ah |
| 20:08.41 | xfighter | I am dying to finish the server before the end of this summer |
| 20:08.50 | xfighter | I neeed to start on my softphone gui |
| 20:08.54 | xfighter | am using java |
| 20:09.05 | xfighter | which is harder than C# as you know |
| 20:09.14 | mgroman | Should I update the zaptel working copy and recompile the zaptel drivers? |
| 20:09.20 | SuPrSluG | ~book |
| 20:09.21 | infobot | [~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook |
| 20:09.37 | xfighter | there will be a website also that will do the same functionality of the gui softohine |
| 20:09.41 | xfighter | softphone* |
| 20:10.09 | xfighter | so my time for the server is very limited actully |
| 20:10.26 | xfighter | it doesn't mean that I spend my whole year doin the server |
| 20:10.36 | CrazyTux | you spend your whole life! |
| 20:10.42 | CrazyTux | :) |
| 20:10.43 | xfighter | I can give it 2 months |
| 20:10.52 | xfighter | :D |
| 20:10.57 | xfighter | ok science is endless |
| 20:11.07 | xfighter | but I want the part that concerns me only |
| 20:11.09 | ariel_ | wow 2 months... |
| 20:11.24 | xfighter | yes ariel no more no less |
| 20:11.32 | xfighter | am talkin about a professional server here |
| 20:11.47 | xfighter | but an asterisk one I guess it'll take less than that |
| 20:11.51 | ariel_ | with some reading you should be able to set the server up within a week or less. At least I did about 6 or 7 years ago with no real doc's back then everyone just said read the code. |
| 20:12.33 | xfighter | aha but the professor has another opinion |
| 20:12.48 | xfighter | he says I must do everything from the scratch |
| 20:12.50 | ariel_ | hint read the boot |
| 20:12.55 | ariel_ | boot/book |
| 20:13.02 | xfighter | I AM READING IT |
| 20:13.10 | xfighter | I finished the first 3 chapters |
| 20:13.11 | ariel_ | yes from scratch it's basic setup |
| 20:13.24 | xfighter | and I applied them |
| 20:13.31 | xfighter | I asked u if I need zaptel |
| 20:13.33 | xfighter | many said no |
| 20:13.41 | [TK]D-Fender | xfighter: NO |
| 20:13.42 | kaldemar | xfighter: then continue through 4-8 and you will know if it works. |
| 20:13.45 | xfighter | since am not connecting the server with the landline network |
| 20:14.00 | xfighter | so now I just need to configure my SIP |
| 20:14.37 | ariel_ | with 2 months you can even take the class from Digium |
| 20:15.20 | xfighter | :D |
| 20:15.32 | xfighter | I like to have a big time to work |
| 20:15.40 | xfighter | makes me feel comfortable |
| 20:15.49 | xfighter | and so I create lovely stuff |
| 20:16.11 | dacs | SuPrSluG: am running asterisk-1.6.0.6 |
| 20:16.22 | xfighter | I even left 8 credits (hours) for my spring just to make sure that I have plenty of time :) |
| 20:17.20 | xfighter | oh guess it's time to start configuring the SIP |
| 20:17.23 | SuPrSluG | dacs: did you have an older version of * ? try moving that dir and do another make install |
| 20:17.24 | xfighter | anyway |
| 20:17.28 | xfighter | may I ask |
| 20:17.28 | ariel_ | dacs: depending on how you follow those doc's but it's missing a few things and it's also using an older version of asterisk. |
| 20:17.31 | cvnet | I really need help testing my h323 to see if it is working, it would be greatly appreciated if you can help me send a test call please help |
| 20:17.33 | xfighter | # mv sip.conf sip.conf.sample |
| 20:17.34 | xfighter | # touch sip.conf |
| 20:17.37 | xfighter | these commands |
| 20:17.47 | xfighter | from which directory should be excecuted? |
| 20:18.10 | cvnet | .. /etc/asterisk/ |
| 20:18.44 | xfighter | the problem of the book is that it doesn't really state the operation field |
| 20:18.57 | ariel_ | cvnet: I wish I could help. But I don't use h323 and don't know very many that due. |
| 20:19.13 | CrazyTux | xfighter: do you come from a programming background? |
| 20:19.14 | cvnet | im soo screwed |
| 20:19.23 | *** join/#asterisk M1s3ry (n=M1s3ry@boromir.api-digital.com) |
| 20:19.45 | xfighter | yes |
| 20:19.46 | CrazyTux | xfighter: if so you should be able to dive in and plug and play w/ asterisk if your just now learning it? It's not as hard as one may think. |
| 20:19.48 | xfighter | C++ and JAVA |
| 20:20.20 | xfighter | the problem is that am also new to linux |
| 20:20.29 | xfighter | but I am managing |
| 20:20.39 | CrazyTux | ah |
| 20:20.40 | xfighter | I see it like windows nowadays no big deal |
| 20:20.57 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 20:21.05 | xfighter | it's just a totally new environment and am challenging everything really |
| 20:21.53 | cvnet | ariel_: do you think u can help me set up a ipsec using sip? |
| 20:25.55 | dacs | ariel_: SuPrSluG : i had 1.4. how can i remove that ? |
| 20:26.49 | SuPrSluG | dacs: mv /usr/lib/asterisk/modules/ /usr/lib/asterisk/modules-original |
| 20:27.58 | SuPrSluG | dacs: then just go back to the * source dir and do a make install again |
| 20:30.01 | [TK]D-Fender | checkout time, bbiab |
| 20:31.23 | dacs | SuPrSluG: doing that right now. |
| 20:32.01 | dacs | SuPrSluG: still same thing :( |
| 20:32.47 | xfighter | hi again |
| 20:33.00 | xfighter | shall I restart asterisk after modifyinh sip.conf? |
| 20:33.11 | xfighter | modifying* |
| 20:34.02 | kaldemar | a sip reload in cli will do unless you modified bindaddress or port. |
| 20:34.39 | xfighter | sip reload? |
| 20:34.48 | xfighter | I guess I must continue reading |
| 20:34.56 | SuPrSluG | dacs: for now try adding noload => res_clioriginate.so to /etc/asterisk/modules.conf then try starting asterisk |
| 20:35.01 | *** join/#asterisk chandoo (n=chandoo@ool-4353b978.dyn.optonline.net) |
| 20:35.03 | cvnet | sip reload would do the job |
| 20:35.25 | xfighter | how do I reload the sip? |
| 20:35.45 | dacs | SuPrSluG: how do i do the no load please |
| 20:35.57 | dacs | SuPrSluG: nm |
| 20:37.06 | ariel_ | xfighter: in the cli: sip reload is the command |
| 20:37.24 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 20:37.41 | ariel_ | dacs: modules.conf you can add that line. |
| 20:37.48 | elitecoder | I have several DIDs that I'd like to point to a single extension so I don't need to duplicate the same dialplan over and over. What's a good way to do that? :] |
| 20:38.15 | ariel_ | dacs: SuPrSluGdacs: for now try adding noload => res_clioriginate.so to /etc/asterisk/modules.conf then try starting asterisk |
| 20:38.45 | ariel_ | elitecoder: pattern matching |
| 20:39.09 | xfighter | one more question |
| 20:39.13 | SuPrSluG | but be careful |
| 20:40.04 | xfighter | now what exactly shall I input in my sip.conf |
| 20:40.06 | xfighter | [general] |
| 20:40.06 | xfighter | context=default |
| 20:40.06 | xfighter | allowoverlap=no |
| 20:40.06 | xfighter | bindport=5060 |
| 20:40.07 | xfighter | bindaddr=0.0.0.0 |
| 20:40.07 | xfighter | srvlookup=yes |
| 20:40.09 | xfighter | [authentication] |
| 20:40.11 | xfighter | ??? |
| 20:40.11 | dacs | how do i restart * |
| 20:40.20 | elitecoder | ariel_: well, alright. :] |
| 20:40.28 | dacs | /etc/init.d/asterisk restart |
| 20:40.45 | ariel_ | dacs: since it did not start the first time asterisk -vvvgc |
| 20:40.58 | xfighter | # service asterisk reload |
| 20:41.16 | dacs | got it to work |
| 20:41.19 | ariel_ | xfighter: he is testing and needs to make sure it's running |
| 20:41.21 | dacs | thank you gus |
| 20:41.43 | dacs | SuPrSluG: i will be back |
| 20:41.50 | xfighter | ariel |
| 20:41.57 | xfighter | am confused with sip.conf |
| 20:42.07 | xfighter | I am not gettin the idea here |
| 20:42.13 | xfighter | the book confuses me |
| 20:42.21 | *** join/#asterisk bmg505 (n=leon@196-209-235-22-rndf-esr-5.dynamic.isadsl.co.za) |
| 20:42.22 | xfighter | I dun get the idea |
| 20:42.22 | ariel_ | sip is fairly simple |
| 20:42.29 | dacs | xfighter: i 2nd that |
| 20:42.35 | dacs | bb in a few |
| 20:42.55 | SuPrSluG | dacs: now the fun begins. |
| 20:43.13 | ariel_ | he left.. |
| 20:43.26 | xfighter | what fun |
| 20:43.41 | ariel_ | but xfighter don't post so much here but ask a question about it and lets see if we can help you understand it |
| 20:43.58 | xfighter | good lets start with sip.conf |
| 20:44.06 | ariel_ | if you need to paste use pastebin.ca for more then 2 or 3 lines |
| 20:44.13 | xfighter | make me understand the mechanism |
| 20:44.21 | xfighter | what should I fill it wiht? |
| 20:44.39 | ariel_ | xfighter: have you seen the sample files? |
| 20:44.54 | xfighter | what sample files? |
| 20:45.42 | ariel_ | backup your files, in the /etc/asterisk |
| 20:46.01 | ariel_ | and from the /usr/src/asterisk do make samples |
| 20:46.34 | ariel_ | all the samples have comments on them as to what there used for |
| 20:48.39 | xfighter | whats the difference between sip.conf and sip.conf.sample?? |
| 20:48.54 | ariel_ | sample |
| 20:48.58 | ariel_ | education |
| 20:49.00 | ariel_ | reading |
| 20:49.21 | xfighter | so I can modify that and use it as a sip.conf file? |
| 20:49.47 | xfighter | after reanming it I mean |
| 20:49.53 | xfighter | renaming* |
| 20:50.38 | xfighter | no? |
| 20:50.38 | ariel_ | yes |
| 20:51.04 | xfighter | but the sample is very big and there are lots of options |
| 20:51.12 | xfighter | must I modify all of that??! |
| 20:51.35 | ariel_ | I remove them options after I know which work. |
| 20:51.55 | ariel_ | <PROTECTED> |
| 20:52.18 | xfighter | aha so everything else is explained in the book? |
| 20:52.49 | elitecoder | Could anyone tell me what's wrong with this pattern? _+180882041[23]X |
| 20:52.56 | ariel_ | everything well, not everything, but a good start |
| 20:53.21 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 20:53.38 | xfighter | so how should I know? |
| 20:53.38 | ariel_ | how many digits |
| 20:53.55 | elitecoder | 10 |
| 20:53.57 | ariel_ | time and doing things will show it's ways |
| 20:54.00 | ariel_ | then no |
| 20:54.15 | elitecoder | wait 11, for the extra 1 |
| 20:54.21 | ariel_ | your trying to match the last 2 nd 3 |
| 20:54.30 | elitecoder | this is how #'s are sent from bandwidth.com |
| 20:54.37 | elitecoder | Yeah |
| 20:54.38 | *** join/#asterisk simNIX (n=simNIX@156-60.bbned.dsl.internl.net) |
| 20:54.39 | ariel_ | with the plus |
| 20:54.55 | elitecoder | well the plus is just a plus not a # |
| 20:54.57 | elitecoder | literal + |
| 20:55.25 | ariel_ | then you can just do +180882041X,blah |
| 20:56.06 | SuPrSluG | elitecoder:that an e164 format, you don't need the + |
| 20:56.24 | elitecoder | oh wtf the manual lead me to believe all patterns must begin with _ |
| 20:56.25 | ariel_ | or instead of the x [23] |
| 20:56.35 | elitecoder | "an extension name is a pattern if it starts with the underscore symbol (_)." |
| 20:57.26 | SuPrSluG | elitecoder:and your matching 11 digits w/ the X at the end |
| 20:58.02 | ariel_ | so do _+180882041[23],blah |
| 20:58.29 | ariel_ | I still don't understand them sending the + |
| 20:58.45 | elitecoder | ok the 23 is a single number |
| 20:58.47 | elitecoder | a 2 or a 3 |
| 20:58.52 | elitecoder | followed by any number 0-9 |
| 20:59.17 | SuPrSluG | yes, that's how it'll be interpreted |
| 20:59.34 | elitecoder | that's what I want I don't get why it's not matching |
| 21:00.40 | elitecoder | Well I'll keep messing with it |
| 21:00.55 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 21:00.57 | elitecoder | thanks guys |
| 21:01.06 | SuPrSluG | cya |
| 21:01.53 | ariel_ | elitecoder: you can use NoOp and a _X. to see what there sending you |
| 21:02.03 | elitecoder | I know what they're sending |
| 21:02.24 | elitecoder | +18088204124 |
| 21:03.00 | elitecoder | the 2 at the end might be a 3, and the last digit could be anything cause we have 10 #'s |
| 21:03.16 | kaldemar | _X. won't match if they really send +. |
| 21:03.36 | ariel_ | yep it would be _. |
| 21:03.39 | *** join/#asterisk justdave (n=dave@unaffiliated/justdave) |
| 21:04.35 | elitecoder | k it works now... god knows why |
| 21:04.39 | elitecoder | did the same thing i've been doing |
| 21:04.43 | elitecoder | reloaded dialplan etc |
| 21:04.56 | marv[work] | I have a queue with a timeout of 7. I see in the queue long some RINGNOANSWER's with 7000 as the ring time, which makes sense. However, I also see some of 60000, which makes no sense. |
| 21:04.59 | elitecoder | I use _+. |
| 21:06.33 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:07.05 | marv[work] | s/queue long/queue log/g |
| 21:07.39 | xfighter | ariel |
| 21:08.08 | xfighter | do I have to configure the sip.conf or is it well configured by asterisk? |
| 21:08.24 | [TK]D-Fender | xfighter: you |
| 21:08.38 | xfighter | yes professor |
| 21:08.56 | xfighter | but it looks complicated |
| 21:09.19 | stope | 486 response from linksys phone, not on DND mode, what needs altering? |
| 21:09.32 | xfighter | at least what are the sections that I necessarily need to modify |
| 21:10.29 | [TK]D-Fender | xfighter: [general] and every peer you configure |
| 21:10.34 | *** join/#asterisk intralanman (n=lanman@173-132-59-225.pools.spcsdns.net) |
| 21:10.55 | xfighter | just the general section |
| 21:10.57 | xfighter | ok |
| 21:11.01 | xfighter | professor |
| 21:11.13 | xfighter | am gonna ask you a couple of question |
| 21:11.18 | xfighter | plz stay around |
| 21:11.59 | xfighter | bindaddr |
| 21:12.05 | xfighter | 0.0.0.0 |
| 21:12.12 | xfighter | shall I modify thsis? |
| 21:12.17 | ariel_ | no |
| 21:12.21 | [TK]D-Fender | nop |
| 21:12.36 | xfighter | :D whats exactly I must modify :D |
| 21:13.03 | ariel_ | rofl, it depends on what you want to do. |
| 21:13.29 | xfighter | right now I just want to see the server working no more no less :D |
| 21:13.38 | xfighter | can I use the settings as they are |
| 21:13.39 | xfighter | ? |
| 21:14.33 | xfighter | ariel professor? |
| 21:15.24 | [TK]D-Fender | xfighter: Go read this guide as a SAMPLE of a simple config, then sit down with the book, get off your ass and go try stuff : |
| 21:15.26 | [TK]D-Fender | ~jerjerguide |
| 21:15.27 | infobot | [~jerjerguide ] Jeremy McNamara's quick sample guide to setting up * : http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/ |
| 21:15.38 | ariel_ | basic is yes, I think that the sample even has some extensions already setup. I have not used those files in a few years. |
| 21:15.38 | elitecoder | heh |
| 21:16.23 | xfighter | professor :D pleeease man , I used to do things this way , first of all make it work secondly start playin around with it |
| 21:16.25 | ariel_ | I think someone here needs a hug or some icecream |
| 21:16.32 | xfighter | I just wanna make the server start |
| 21:16.49 | xfighter | then I will see what can I do about modifications later |
| 21:17.00 | ariel_ | xfighter: you said that asterisk -r worked? |
| 21:17.17 | xfighter | you mean the status? |
| 21:17.21 | ariel_ | if that gives you a cli then it's working.... |
| 21:17.22 | [TK]D-Fender | xfighter: There is no such thing as "work" |
| 21:17.35 | [TK]D-Fender | xfighter: What does "work" mean when you don't configure anything? |
| 21:17.44 | xfighter | it is time to ask u what CLI is |
| 21:18.03 | [TK]D-Fender | ariel_: DFTT |
| 21:18.13 | elitecoder | xfighter: command line interface :o |
| 21:19.01 | xfighter | :D |
| 21:19.06 | xfighter | it's running ariel |
| 21:19.19 | *** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net) |
| 21:19.33 | Ritzerisk | is there a way to create a phamtom Voicemail |
| 21:19.55 | xfighter | connected to asterisk 1.4.25.1 currently running on localhost (pid =2634) |
| 21:20.13 | [TK]D-Fender | Ritzerisk: And since you just asked in FreePBX.... well... whats that supposed to eman to us? |
| 21:20.32 | [TK]D-Fender | xfighter: Congratulations... * is "running". Does that count as "working"? |
| 21:20.51 | xfighter | can I use softphone to test it? |
| 21:21.07 | [TK]D-Fender | xfighter: akin to starting your motor but never putting it into gear or trying to steer |
| 21:21.17 | aurax | i'm getting 407 proxy authentication and 401 unauthorized errors, suddently... everything worked fine.. any idea? |
| 21:21.24 | [TK]D-Fender | xfighter: With what? Youdidn't set it up for use with one now did you? |
| 21:22.01 | xfighter | I dun get it this the part I didn't understand from the book |
| 21:22.24 | [TK]D-Fender | xfighter: You need to configure all sorts of config files or * doesn't do jack shit. |
| 21:23.38 | [TK]D-Fender | xfighter: Soft-phone? * will refuse to talk to ANYTHING until you configure it to. Accept random calls? Why should it? What should you be allowed to dial? What numbers & patterns? Whats it supposed to do? IT is your job to configure ABSOLUTELY EVERYTHING. |
| 21:23.38 | xfighter | :D all of them ??!!!!?! :S |
| 21:24.10 | [TK]D-Fender | xfighter: read the blog post I gave you read the book and get off your ass. This hand holding attempt of yours it just wrong... |
| 21:24.25 | xfighter | jerjer guide? |
| 21:24.28 | aurax | any idea ? |
| 21:24.32 | [TK]D-Fender | xfighter: yes. |
| 21:24.38 | beek | points [TK]D-Fender to the ClueBat (tm) |
| 21:24.52 | [TK]D-Fender | aurax: Auth is wrong if it 401's after a 407 |
| 21:25.12 | stope | 486? |
| 21:26.44 | aurax | [TK]D-Fender: i see only 407's now. |
| 21:27.10 | xfighter | professor is it only at my side or the site is not working? |
| 21:27.17 | xfighter | The page you are looking for is temporarily unavailable. |
| 21:27.18 | xfighter | Please try again later. |
| 21:27.34 | aurax | very weird, just like that... |
| 21:28.02 | aurax | the weird thing is that in sip show registry i see that this trunk is registered.. |
| 21:28.26 | aurax | but when i try to estelish a connection (incoming) i see the 407 pops again |
| 21:28.57 | xfighter | professor? |
| 21:29.03 | aurax | seems like i receive 407 unauthorized, but it suppose to work... it's a net2phone trunk |
| 21:29.26 | *** join/#asterisk Greek-Boy (n=greek@41.222.89.77) |
| 21:29.35 | xfighter | I guess the site you provided is not working |
| 21:30.42 | [TK]D-Fender | aurax: Or you aren't expected to 407 them and allow it to auth by host |
| 21:31.18 | aurax | i see that i receive the 407 from them. |
| 21:31.25 | [TK]D-Fender | xfighter: http://74.125.95.132/search?q=cache:x-SVjZ-02u8J:www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/+http://www.jeremy-mcnamara.com/2007/02/26/how-to-configure-asterisk-your-first-installation/&cd=1&hl=en&ct=clnk&gl=ca |
| 21:31.25 | aurax | is it a reject reponde from my asterisk ? |
| 21:31.35 | *** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com) |
| 21:31.47 | xfighter | thanx professor |
| 21:32.04 | keith4 | would anyone care to share an opinion on Citel products? |
| 21:32.41 | cvnet | Anyone here who uses H323 could help me out with testing a call ? |
| 21:33.32 | rue_mohr | too bad nobody could tell me that echo canceler switch was for all lines and not a per-line setting |
| 21:34.58 | [TK]D-Fender | rue_mohr: it is per channel |
| 21:35.33 | [TK]D-Fender | keith4: somewhat flakey and only suggestable if you're FUBAR'd for wiring and desparate |
| 21:35.40 | [TK]D-Fender | keith4: Pricey |
| 21:36.30 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:36.30 | keith4 | I guess it's not a good sign the Voip Supply only has one product made by them, either |
| 21:36.48 | [TK]D-Fender | keith4: They only make 2 products :) |
| 21:37.13 | keith4 | unless you count the discontinued one |
| 21:37.18 | keith4 | then it's 3! |
| 21:38.34 | keith4 | k, thanks [TK]D-Fender |
| 21:38.45 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 21:39.12 | *** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee) |
| 21:41.54 | cvnet | [TK]D-Fender: how do you control/set/change packetization timer for H323 ? |
| 21:43.08 | [TK]D-Fender | cvnet: What does it TELL you to do? |
| 21:43.32 | cvnet | [TK]D-Fender: what does what tell me? |
| 21:43.45 | elitecoder | Is there a good way to verify the number of channels a provider has assigned to you? |
| 21:44.58 | *** join/#asterisk CrazyTux (n=brandon@ip98-164-236-47.oc.oc.cox.net) |
| 21:45.01 | cvnet | elitecoder: u can ask them, the only way you can test it yourself is to start sending x calls until it no longer connectds |
| 21:45.24 | elitecoder | Alright. |
| 21:45.25 | elitecoder | Is there a common error message you see in the CLI if you've max'd it out? |
| 21:45.31 | CrazyTux | Hey guys -- say I have a goto() method, and the context/extension/label/priority whatever it is going to does not exist, does it just go down the list to my next exten => priority, or do I need to have some kind of invalid handler? |
| 21:45.31 | cvnet | reject i guess |
| 21:46.10 | cvnet | [TK]D-Fender: sorry i didtn get ur answer |
| 21:47.01 | [TK]D-Fender | cvnet: the docs & sample configs |
| 21:47.20 | *** part/#asterisk Errotan (n=Errotan@5403E4F1.catv.pool.telekom.hu) |
| 21:47.40 | [TK]D-Fender | [17:43]<elitecoder>Is there a good way to verify the number of channels a provider has assigned to you? <- Pick up a phone and ask them |
| 21:47.46 | j_kroon | r5842 | sruffell | 2009-01-27 11:00:58 -0600 (Tue, 27 Jan 2009) | 4 lines <-- I'm looking for this single patch so that I can get dahdi-2.1.0.4 compiling against kernel 2.6.29. With ViewVC being down ... what other options do I have to get to it? |
| 21:53.19 | cvnet | [TK]D-Fender: I read the h323.conf again and the onlything i found in regards to timing is (jbmaxsize = 200) now is that the same as packetization timer ? |
| 21:53.57 | *** part/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 21:55.07 | *** join/#asterisk RobH (n=RobH@cpe-173-169-30-118.tampabay.res.rr.com) |
| 21:56.58 | *** join/#asterisk DarkRift (n=dark@65.92.169.244) |
| 21:59.17 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 22:00.02 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.36.70) |
| 22:01.16 | xfighter | peer = voip devices & softphones?? |
| 22:03.38 | *** join/#asterisk arthax0r (i=arthax0r@gateway/shell/blinkenshell.org/x-242f96ae96729d08) |
| 22:03.45 | xfighter | what if I use friend , it can be either peer or user right |
| 22:04.02 | xfighter | would u guyz explain me the different between the three types |
| 22:04.30 | xfighter | difference* |
| 22:05.23 | *** join/#asterisk Aiatek (n=Munoz@190.94.58.21) |
| 22:06.13 | wdoekes | http://lmgtfy.com/?q=asterisk+peer+vs+user |
| 22:08.20 | bmoraca | dude |
| 22:08.28 | bmoraca | Adtran's TA900's are freaking awesome |
| 22:09.24 | xfighter | so whats the use when I define the type of the SIP as peer and then set the source host as dynamic |
| 22:09.50 | xfighter | can I say type=user without defining the host type? |
| 22:11.11 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-206.phlapa.fios.verizon.net) |
| 22:11.39 | xfighter | professor? |
| 22:11.56 | rue_mohr | bmoraca, what are they? |
| 22:12.08 | bmoraca | rue_mohr: media gateway |
| 22:12.13 | rue_mohr | oh |
| 22:12.20 | rue_mohr | opposed to a tdm800? |
| 22:12.28 | rue_mohr | with the level and echo issues? |
| 22:12.35 | bmoraca | router, t1 csu/dsu, analog converter all built in to one |
| 22:12.43 | rue_mohr | ah |
| 22:12.57 | bmoraca | lets me provide PRI and analog service to my customers with legacy PBX systems over a VoIP trunk |
| 22:13.17 | rue_mohr | I just had a customer asking about that this morning |
| 22:14.21 | bmoraca | it was an absolute pain in the freaking balls to get working...but now that it is, it's wonderful |
| 22:24.13 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 22:25.54 | *** join/#asterisk mpeterson (n=cooldown@pepperz.de) |
| 22:27.35 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 22:33.31 | *** join/#asterisk kharsa (n=akharsa@190.244.52.130) |
| 22:34.03 | kharsa | hello, need some help |
| 22:34.58 | kharsa | running trixbox have chan_sip.c: failed to athenticate user "343423423"... from an outside call, any help? |
| 22:36.09 | Aiatek | did you asked in trixbox channel? |
| 22:36.27 | kharsa | didn't know that exists.. switching, thank you |
| 22:39.29 | kharsa | calling from outside gets busy tone everytime, i have some problem with my trunk configuration maybe... |
| 22:40.39 | *** join/#asterisk Iamnach0 (i=Iamnacho@ip68-103-153-140.ks.ok.cox.net) |
| 22:43.48 | *** join/#asterisk war9407 (i=war@liquidswords.org) [NETSPLIT VICTIM] |
| 22:43.48 | *** join/#asterisk ldsjohn (n=jbunn@c-98-246-83-83.hsd1.wa.comcast.net) [NETSPLIT VICTIM] |
| 22:43.48 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) [NETSPLIT VICTIM] |
| 22:43.48 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM] |
| 22:43.49 | rue_mohr | so in order for page to work I have to set a sip-alert to get the phones to auto pick up |
| 22:46.43 | *** join/#asterisk layne (n=layne@ool-44c0048f.dyn.optonline.net) |
| 22:46.51 | kharsa | hello guys, how do I properly configure my trunks to a SIP service in trixbox... I can call outside but calling from outside get busy tone and Failed to Authentica user error |
| 22:47.36 | russellb | kharsa: you should try AsteriskNOW instead of trixbox :-) |
| 22:47.40 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 22:47.49 | russellb | in either case, not many people work with FreePBX in here. |
| 22:48.57 | Aiatek | <kharsa> you need to define an incoming route in your FreePbx |
| 22:49.09 | kharsa | thank you for the advise but my boss wont be very happy if I do that |
| 22:49.12 | Aiatek | but the rest is up to you |
| 22:49.27 | kharsa | aitatek: I did that, but it isnt workin |
| 22:50.55 | *** join/#asterisk rvhi (n=chatzill@207.2.110.6) |
| 22:50.57 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 22:51.59 | rvhi | i have an analog adapter for fax machine, not reliable even with g711. Someone suggests using t38. any commercial product for that? too much work to configure it |
| 22:53.15 | rue_mohr | you should use ulaw or alaw |
| 22:53.29 | rue_mohr | ? |
| 22:54.50 | carrar | "too much work to configure it" |
| 22:54.52 | carrar | wtf |
| 22:55.17 | carrar | Sounds like you need to just buy a POTS line and be done with it |
| 22:55.37 | carrar | otherwise it's "too much work to configure it" |
| 22:56.13 | rvhi | if you don't have a suggestion, then don't say it. wtf! |
| 22:56.29 | rue_mohr | why dont you set it to ulaw |
| 22:56.42 | rue_mohr | then again, maybe the problem is jitter |
| 22:56.50 | rue_mohr | whats the network like |
| 22:56.54 | rvhi | ulaw is the same as g711 |
| 22:57.07 | rvhi | it works sometimes |
| 22:57.12 | rvhi | but not reliable |
| 22:57.19 | rue_mohr | stares blankly at the screen |
| 22:57.42 | drmessano | POTS line <-- reliable commercial fax product |
| 22:58.40 | rue_mohr | hmm, k |
| 22:59.00 | Nugget | drmessano speaks the truth |
| 22:59.02 | rue_mohr | so, you need to know where you problem is |
| 22:59.06 | rvhi | POTS costs $50 per month |
| 22:59.13 | rvhi | $600 per year |
| 22:59.15 | rue_mohr | your using the adapter with a pri? |
| 22:59.18 | *** join/#asterisk ctp (n=ctp@brsg-d9bee4e9.pool.mediaWays.net) |
| 22:59.23 | rue_mohr | we pay $80 here |
| 22:59.41 | rue_mohr | is you service via pri? |
| 22:59.54 | rue_mohr | are you using a T1 for your co connection? |
| 23:00.01 | rvhi | fax machine goes to adapter, via IP network, to a server with PRI lines |
| 23:00.08 | rue_mohr | aha! |
| 23:00.23 | rvhi | i want to enable t38 from adapter to the server |
| 23:00.29 | lancey | asterisk can forward t.38 if the ata and the server with the pri lines can do it |
| 23:00.29 | rue_mohr | so, what do you know about pri synchronization? |
| 23:00.38 | rvhi | the server handles VOIP calls |
| 23:00.39 | rue_mohr | what is the clock source for your pri? |
| 23:00.57 | rvhi | from carrier |
| 23:01.01 | rue_mohr | k |
| 23:01.13 | rue_mohr | runs out of questions |
| 23:01.16 | rvhi | the issue is IP network, public internet |
| 23:01.17 | lancey | erm.. voip or pri? |
| 23:01.26 | rue_mohr | jitter |
| 23:01.31 | rue_mohr | up your jitter buffer |
| 23:01.40 | rvhi | t38 is supposed to fix the problem |
| 23:01.48 | rue_mohr | jitter buffer |
| 23:02.14 | rvhi | i thought fax can't handle the jitter buffer |
| 23:02.18 | rvhi | but anyway, i can try it |
| 23:02.25 | rvhi | how to enable jitter buffer? |
| 23:03.51 | Aiatek | sip or iax jitter buffer |
| 23:03.52 | Aiatek | ? |
| 23:05.20 | drmessano | How is the public internet a factor here? |
| 23:05.31 | drmessano | You said you had an adapter running to some server with a PRI |
| 23:06.35 | lancey | thinks it's high time everyone learns to use e-mail instead of fax |
| 23:06.44 | *** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net) |
| 23:07.32 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 23:09.55 | *** join/#asterisk elitecoder (n=liq@apollo.bullethost.com) |
| 23:10.36 | elitecoder | Is there a way to see how many channels are active for a sip provider? |
| 23:10.46 | elitecoder | like a cli command |
| 23:12.42 | elitecoder | show channels: Display information on channels maybe |
| 23:13.00 | rue_mohr | hey aastra has the ability to monitor an rtp port for pages |
| 23:13.18 | rue_mohr | why the hell dosn't polycom have that, polycom sucks... |
| 23:17.43 | *** part/#asterisk kharsa (n=akharsa@190.244.52.130) |
| 23:19.20 | rue_mohr | how would I make use of that, an rtp port that is monitored for phone pages? |
| 23:20.40 | rue_mohr | the only ways I can see to do the paging, it sounds like it would take over the handset of anyone on the phone |
| 23:20.57 | rue_mohr | and NO I cant just try that with the office open |
| 23:23.40 | *** join/#asterisk blkry (n=blkry@97.95.233.232) |
| 23:31.23 | rue_mohr | -help- |
| 23:31.31 | rue_mohr | my new install disn't load the dahdi |
| 23:32.25 | rue_mohr | shoudln't core show channeltypes list dahdi |
| 23:32.26 | rue_mohr | ? |
| 23:33.04 | rue_mohr | where is my dahdi device? |
| 23:33.26 | rue_mohr | root@debian:/usr/src# dahdi_monitor 1 |
| 23:33.27 | rue_mohr | Nothing to do with the stream(s) ... |
| 23:33.29 | rue_mohr | whats that mean? |
| 23:34.01 | rue_mohr | [Jun 9 16:32:58] WARNING[5318]: channel.c:3210 ast_request: No channel type registered for 'dahdi' |
| 23:34.04 | rue_mohr | whats this |
| 23:34.07 | rue_mohr | why didn't this work |
| 23:34.16 | rue_mohr | did I forget something when I compiled asterisk? |
| 23:34.37 | M1s3ry | did you compile dahdi? |
| 23:34.59 | M1s3ry | <PROTECTED> |
| 23:35.09 | rue_mohr | yes |
| 23:35.21 | rue_mohr | make ; make install |
| 23:35.32 | rue_mohr | and confgure |
| 23:35.53 | rue_mohr | the kernel driver isloaded |
| 23:36.00 | M1s3ry | Did you use dahdi linux and source, or complete? |
| 23:36.20 | rue_mohr | asterisk-1.4.25.1 dahdi-linux-complete-2.1.0.4+2.1.0.2 |
| 23:36.22 | rue_mohr | I compiled both |
| 23:37.09 | rue_mohr | didn't I have this problem before where it wouldn't work right with dahdi |
| 23:37.31 | lancey | has 1.4 moved to dahdi? |
| 23:37.46 | lancey | o_O |
| 23:37.48 | rue_mohr | I was just using 1.4.22, upped it to 1.4.25 |
| 23:38.13 | rue_mohr | in make menuconfig, chan_dahdi has a *, what does that mean? |
| 23:39.25 | rue_mohr | it must mean Y |
| 23:40.36 | rue_mohr | drat this is my only window to do this work |
| 23:41.18 | rue_mohr | [Jun 9 16:29:06] WARNING[5285] chan_dahdi.c: Unable to specify channel 1: No such device or address |
| 23:42.51 | rue_mohr | WHAT device or address is it LOOKING for!?!?!?! |
| 23:43.48 | rue_mohr | ok, dahdi_monitor does work |
| 23:43.51 | rue_mohr | so whats wrong |
| 23:43.52 | carrar | woah, might have to the docs in the software you compiled |
| 23:43.58 | carrar | heavens no |
| 23:43.59 | *** join/#asterisk AlmightyOatmeal (n=jamie@66-190-59-171.dhcp.mdsn.wi.charter.com) |
| 23:44.06 | rue_mohr | my old compile worked |
| 23:44.51 | carrar | to ^read |
| 23:44.51 | rue_mohr | but asterisk needs to load a dahdi thing |
| 23:45.00 | rue_mohr | read whcih? |
| 23:45.19 | carrar | You tell me |
| 23:45.37 | carrar | </handholding> |
| 23:45.54 | rue_mohr | I dont get how asterisk knows where the dahdi source is |
| 23:45.56 | AlmightyOatmeal | i forgot.. where does one set the mailbox password for a specific user? |
| 23:46.12 | carrar | voicemail.conf |
| 23:46.33 | AlmightyOatmeal | silly me :) |
| 23:46.36 | carrar | yes |
| 23:46.49 | AlmightyOatmeal | tuesday, its the new monday :P |
| 23:46.55 | rue_mohr | make menuconfig has dadhi turned on |
| 23:48.09 | rue_mohr | what changed after asterisk 1.4.22 thats got this broken |
| 23:48.32 | rue_mohr | /etc/asterisk/chan_dahdi.conf unless 'dahdichanname' has been set to |
| 23:48.32 | rue_mohr | <PROTECTED> |
| 23:48.52 | carrar | s/what changed after asterisk 1.4.22 thats got this broken/What did I not pay attention to so that I am way out of clue/ |
| 23:49.28 | rue_mohr | ;dahdichanname = yes ; Channels created by chan_dahdi will be called 'DAHDI', otherwise 'Zap' <-- asterisk.conf? |
| 23:50.43 | rue_mohr | even with that yes asterisk core show channeltypes dosn't come up with dahdi |
| 23:51.11 | rue_mohr | [Jun 9 16:49:44] ERROR[5453] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory |
| 23:51.19 | *** part/#asterisk ruben23 (n=RPL@124.107.3.178) |
| 23:51.20 | rue_mohr | whats supposed to be in there, why didn't make install create it |
| 23:51.35 | rue_mohr | is it supposed to be a file or dir? |
| 23:52.00 | rue_mohr | who creates that |
| 23:52.04 | rue_mohr | whats the device number? |
| 23:52.05 | lancey | lol |
| 23:52.20 | rue_mohr | is it a driver I missed? |
| 23:52.34 | rue_mohr | sholdn't the dahdi source have done this for me? |
| 23:52.36 | carrar | someone kill that run away bot |
| 23:52.44 | rue_mohr | hey i need to get this back up! |
| 23:52.49 | rue_mohr | its the office phone system! |
| 23:53.10 | rue_mohr | I appreciate your not working with a live system that people are depending on, but I am |
| 23:53.13 | carrar | guess you shouldn't be messing with a live production system if you don't know what you're doing |
| 23:53.26 | rue_mohr | and how the hell am i supposed to get there?! |
| 23:53.37 | carrar | on a none production system |
| 23:53.45 | carrar | so you can test and learn |
| 23:53.52 | rue_mohr | in all that spare time I have eh? |
| 23:53.58 | lancey | o_O |
| 23:53.59 | carrar | You make time |
| 23:54.04 | carrar | like everyone else does |
| 23:54.05 | rue_mohr | I been working with my home system for a few years, its fine |
| 23:54.08 | lancey | rue_mohr: what's in .25 that's not in .22 and you need it? |
| 23:54.11 | rue_mohr | but its not this system |
| 23:54.20 | AlmightyOatmeal | rue_mohr: wow, you are a real douche.. no offense |
| 23:54.27 | rue_mohr | it only costs about $1500 to 'play' with a system, why dosn't everyone do it |
| 23:54.40 | rue_mohr | nobody seems to be able to help me out here |
| 23:54.50 | rue_mohr | I dont want to have to call digium for every problem I have |
| 23:54.54 | carrar | because you're a loose cannon |
| 23:54.56 | AlmightyOatmeal | i think its more of you refuse to help yourself |
| 23:55.04 | AlmightyOatmeal | which is why nobody here is helping you |
| 23:55.15 | rue_mohr | can you help me diagnose why asterisk cant load the dahdi channels? |
| 23:55.21 | rue_mohr | I'm freaking panicing |
| 23:55.25 | rue_mohr | giveme a break |
| 23:55.46 | AlmightyOatmeal | read the manual :) |
| 23:55.46 | rue_mohr | I have negitive 30 mins to get this back up |
| 23:55.48 | carrar | better call support |
| 23:55.48 | rue_mohr | I did, says nothing about this |
| 23:55.51 | AlmightyOatmeal | undo whatever you messed up |
| 23:55.58 | carrar | no shit :) |
| 23:56.05 | AlmightyOatmeal | BACKUP YOUR CONFIG FILES! |
| 23:56.08 | AlmightyOatmeal | oops </common sense> |
| 23:56.10 | AlmightyOatmeal | :) |
| 23:56.12 | carrar | use your backout plan |
| 23:56.13 | rue_mohr | already did |
| 23:56.19 | rue_mohr | not the point |
| 23:56.29 | rue_mohr | the fax machine didn't work anyhow |
| 23:56.35 | AlmightyOatmeal | sighs |
| 23:56.38 | carrar | heh |
| 23:56.40 | rue_mohr | the polycom phones suck... |
| 23:56.45 | rue_mohr | people hate this system |
| 23:56.54 | carrar | sounds like a FAILED install |
| 23:56.55 | rue_mohr | they hate asterisk because of this system |
| 23:56.59 | rue_mohr | yea, |
| 23:57.01 | lancey | lol |
| 23:57.03 | carrar | 3 steps |
| 23:57.06 | carrar | 1 FIX |
| 23:57.07 | carrar | 2 IT |
| 23:57.11 | carrar | 3 FIX IT |
| 23:57.11 | lancey | do they know what *is* asterisk? |
| 23:57.15 | AlmightyOatmeal | they should hate asterisk because of YOU not asterisk itself |
| 23:57.20 | rue_mohr | yes, |
| 23:57.34 | rue_mohr | no they know I'm workinghard to make this do what they want |
| 23:57.37 | carrar | they should hate you for installing something you can't support! |
| 23:57.43 | AlmightyOatmeal | quite |
| 23:57.48 | rue_mohr | I'v done alot independant of the cahnnels help |
| 23:58.06 | rue_mohr | my system at hole is all analog, T1 to channelbank, sinch |
| 23:58.12 | carrar | grea |
| 23:58.14 | carrar | t |
| 23:58.20 | carrar | bring in your home system |
| 23:58.23 | carrar | if it works |
| 23:58.27 | rue_mohr | if tk hadn't told me to get these freaking polycom sets I'd been finished long ago |
| 23:58.36 | carrar | polycom is the best |
| 23:58.39 | carrar | and hella fast |
| 23:58.41 | rue_mohr | NO ITS NOT |
| 23:58.47 | lancey | polycom rules, what's wrong with them? |
| 23:58.48 | rue_mohr | POLYCOM SUCKS |
| 23:58.48 | carrar | thats odd |
| 23:58.54 | rue_mohr | they cant do proper paging |
| 23:58.57 | rue_mohr | the support sucks |
| 23:58.57 | carrar | Must be polycom |
| 23:59.00 | carrar | thats crazy |
| 23:59.03 | rue_mohr | the dtmf is screwed |
| 23:59.05 | carrar | heh |
| 23:59.12 | rue_mohr | and they are short enough buttons for a proper office touse |
| 23:59.16 | carrar | Did you configure the DTMF secion of the xml config? |
| 23:59.22 | rue_mohr | and they take up too much desk space |
| 23:59.29 | lancey | meh, that's just bullshitting for me. |
| 23:59.40 | carrar | Did you setup Asterisk with some DTMF switches switched? |
| 23:59.56 | rue_mohr | carrar, if they properly described the controls in the polycom manual it would help |