| 00:00.27 | [TK]D-Fender | ambush276: Good, now you need to find a mode that is compatible. I'd try adding "dtmfmode=inband" first to test |
| 00:00.48 | ambush276 | so do that under teh [12345] |
| 00:00.51 | ambush276 | in teh sip.conf? |
| 00:01.55 | *** join/#asterisk awkfu (n=awkfu@166.189.62.81) |
| 00:02.18 | ambush276 | http://pastebin.ca/1404248 |
| 00:06.01 | [TK]D-Fender | ambush276: Do as it says then... and I'd recommend not using GSM |
| 00:06.09 | ambush276 | how do i change that? |
| 00:06.10 | *** join/#asterisk qdk (n=qdk@0x55814435.terminal.tdcmobil.dk) |
| 00:10.13 | [TK]D-Fender | change "inband" to "rfc2833" LIKE IT SAYS |
| 00:14.28 | ambush276 | kk |
| 00:17.40 | ambush276 | [TK]D-Fender: where do i change teh codec? is it in SIp.conf? |
| 00:17.46 | ambush276 | there is no command for inband=rfc2833 |
| 00:20.20 | [TK]D-Fender | [20:00]<[TK]D-Fender>ambush276: Good, now you need to find a mode that is compatible. I'd try adding "dtmfmode=inband" first to test |
| 00:20.27 | [TK]D-Fender | [20:10]<[TK]D-Fender>change "inband" to "rfc2833" LIKE IT SAYS |
| 00:20.40 | [TK]D-Fender | ambush276: Are there any functioning neurons in there at all? |
| 00:21.06 | [TK]D-Fender | thinks the lights are on, the wheel is spinning, but ambush276's hamster is FUCKING DEAD |
| 00:23.42 | ambush276 | http://pastebin.ca/1404262 |
| 00:23.47 | ambush276 | that is what kicks back |
| 00:23.54 | ambush276 | and still same problem w/ dialing numbers |
| 00:26.24 | [TK]D-Fender | ambush276: add "disallow=all", "allow= ulaw" (in THAT order) to your peer, and change the dtmfmode back to "inband" |
| 00:26.55 | ambush276 | kk one sc. |
| 00:27.25 | Slade- | hey fastagi operates asynchronously on the asterisk side right? like if i take 50 seconds to respond to a specific request the remaining requests will continue on.. correct? |
| 00:29.18 | ambush276 | it worked ! thanks |
| 00:29.21 | ambush276 | btw might i ask |
| 00:29.35 | ambush276 | why did the disallow=all and allow=ulaw work? what did it do to ths system? |
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| 00:36.25 | [TK]D-Fender | ambush276: Changed the codec used for the call to one that supports inband DTMF and hoped that's how it would be passed on. |
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| 01:00.57 | majorriley | [TK]: if you are there thanks a bungs. Fax works great!!! |
| 01:01.07 | majorriley | bunch |
| 01:01.55 | [TK]D-Fender | majorriley: You're welcome |
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| 01:26.44 | [TK]D-Fender | "hosted.ap.org â The U.S. declared a public health emergency Sunday to deal with the emerging new swine flu, much like the government does to prepare for approaching hurricanes.More⦠(World News)" |
| 01:26.46 | [TK]D-Fender | LOL! |
| 01:26.49 | [TK]D-Fender | Katrina 2.0! |
| 01:27.12 | [TK]D-Fender | throws another sand-bag at the pigs |
| 01:27.36 | drmessano | lol |
| 01:29.28 | KyleK | aww man |
| 01:30.02 | KyleK | i need to buy something from a store in a few days, hopefully this blows over |
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| 01:47.27 | drmessano | Qwell: |
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| 03:11.55 | Sapote | anybody know like as FOP Flash Operator Panel? for control of many rental telephone cabins. |
| 03:16.54 | kc8pxy | Sapote: telephone "cabins" ?? |
| 03:18.17 | jameswf | telephones have their own cabins |
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| 03:26.36 | Sapote | clear |
| 03:27.04 | Sapote | :D |
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| 04:45.08 | omer | hello |
| 04:45.45 | omer | ... /usr/bin/ld: cannot find -lssl <------ I get this message when i do "make" to asterisk...any help? |
| 04:45.58 | omer | which package I need to fix this |
| 04:47.29 | jameswf | libssl maybe? |
| 04:48.00 | omer | or openssl?> |
| 04:49.30 | jameswf | omer: http://tinyurl.com/cuft3z |
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| 04:52.55 | omer | ok let me check. |
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| 05:16.13 | omer | jamesswf: which is better yum or apt-get? |
| 05:16.23 | omer | is package name apt-get? or what? |
| 05:16.55 | omer | my yum is always giving HTTP error 404 ... :S |
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| 05:31.13 | shyam_k | now that i have dahdi-kernel and dahdi-tools installed but the kernel module isn't loading prolly as there is those zaptel module .. but its depending upon a bunch of other modules wc* xpp tor etc. should i be editing some init files to make it sure zaptel won't load next time? |
| 05:32.59 | shyam_k | should i be removing /etc/init.d/zaptel? |
| 05:39.16 | omer | getting this message on asterisk "make" ..... http://www.pastebin.ca/1404509 |
| 05:39.18 | omer | any help? |
| 05:51.12 | shyam_k | okay am done with that.. whats lszaptel equivalent of dahdi? anyway am done with cat /proc/dahdi/1 |
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| 06:40.32 | shyam_k | ha. after going through all those to move from zaptel to dahdi now it says it can't find zapata.conf?! |
| 06:42.50 | shyam_k | okay i could ignore that right:) |
| 06:43.23 | omer | I am really stucked in compiling Asterisk-1.2.14 in RH9 ..... http://www.pastebin.ca/1404545 |
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| 06:58.58 | shyam_k | is there anyway to reload every module related to dahdi? or i have to go handpick each module and finally dahdi module? |
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| 07:56.22 | jbjuly | I've written a custom meetme app, I've found out that Admin and User pincode is useless since no Users can join the conference if Admin has already joined. I'm wondering if anyone know a workaround other than adding an admin flag to User. |
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| 08:00.51 | jbjuly | Hi, does anyone know a workaround regarding the default MeetMe behaviour, that when an Admin joined the conference, the Users trying to join will be flashed with an invalid PIN. |
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| 08:34.37 | st_ignucius | i saw module dahdi not found as an error during bootup message.. |
| 08:36.20 | st_ignucius | later as i try ls -l /usr/lib/asterisk/modules/chan_dahdi.so it says -rwxr-xr-x 1 root root 539070 2009-04-26 22:16 /usr/lib/asterisk/modules/chan_dahdi.so but then as i do modprobe /usr/lib/asterisk/modules/chan_dahdi.so it says module not found!! wht the heck is that?! |
| 08:37.15 | st_ignucius | i don't have /dev/dahdi |
| 08:37.27 | st_ignucius | but i have /dev/zap |
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| 08:50.18 | agx | morning, is Microsoft Communication Server SIP Based? i mean: can i connect it to asterisk using a SIP trunk? |
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| 09:57.44 | freh | st_ignucius, are you using dahdi? |
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| 10:14.05 | markit | hi, I've * behind NAT and also in DMZ, while extensions are in "green" interface. Setting nat parameters in sip.conf makes sip extensions loose voice. Any info/tips about this setup? |
| 10:14.25 | markit | in localnet I've specified DMZ subnet |
| 10:17.04 | markit | ok, I try with multiple localnet |
| 10:17.51 | *** join/#asterisk shyam_k (n=user@unaffiliated/shyam-k/x-8459115) |
| 10:19.16 | markit | works :) thanks, byt |
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| 10:31.51 | Coldfire_ | Hi guys. I begin with Asterisk but already problems ! My Asterisk server is configured with 3 clients but I can't reach them each other. However each clients obtain the welcome message when I compose"3". Does someone has an explication ? |
| 10:35.05 | iamy_china | Did you set the dial plan? |
| 10:35.49 | Coldfire_ | I'm not sure, what does it mean ? |
| 10:36.04 | MaliutaLap | explication ... I'm going to use that one |
| 10:36.19 | iamy_china | You are beginer, right? |
| 10:36.24 | Coldfire_ | yes |
| 10:38.08 | Coldfire_ | MaliutaLap : sorry for that ;) |
| 10:38.48 | MaliutaLap | Coldfire_: do you have a dialplan? (extensions.conf) |
| 10:38.56 | Coldfire_ | oh yes ! |
| 10:39.25 | MaliutaLap | pastebin it and your sip.conf |
| 10:39.29 | MaliutaLap | ~pb |
| 10:39.30 | 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/ |
| 10:39.44 | MaliutaLap | oh, and have you read the book? |
| 10:39.57 | Coldfire_ | just few lines actually .. |
| 10:40.12 | MaliutaLap | ~book |
| 10:40.12 | 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 |
| 10:40.24 | MaliutaLap | pastebin them |
| 10:41.08 | iamy_china | http://www.google.com/url?sa=t&source=custom&ct=res&cd=1&url=http%3A%2F%2Fdownloads.oreilly.com%2Fbooks%2F9780596510480.pdf&ei=nIv1SbfxBpSVkAWNsZjXCg&usg=AFQjCNHH-gNcCSeBwE6DwUNG76JrV64qaw |
| 10:41.27 | iamy_china | Read this book first please |
| 10:42.43 | MaliutaLap | iamy_china: just use ~book |
| 10:43.32 | iamy_china | MaliutaLap: ? |
| 10:43.39 | iamy_china | MaliutaLap: what you mean? |
| 10:44.05 | joobie | guys having an issue with queues.. i've setup one with dynamic members.. when I put the members on pause, if a new call comes in it doesn't allow them to join the queue.. i set the variable 'joinempty=yes' and 'leavewhenempty=yes' .. still no joy. any ideas? |
| 10:44.09 | *** join/#asterisk propellerhead (n=yogurt2u@190.227.250.31) |
| 10:44.28 | MaliutaLap | ~book |
| 10:44.28 | 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 |
| 10:44.37 | MaliutaLap | iamy_china: that's what I mean |
| 10:45.33 | Coldfire_ | Just want to add that with my SIP client I get this error message : " Register attempt for proxy sip:556@IP:port failed" and after "404 Not Found" :D |
| 10:46.06 | Coldfire_ | But I got the welcome message all the same ... |
| 10:46.28 | MaliutaLap | Coldfire_: sip.conf and extensions.conf in pb |
| 10:46.39 | MaliutaLap | Coldfire_: can't help you unless we can see them |
| 10:46.51 | Coldfire_ | yeah I make that |
| 10:49.12 | Coldfire_ | here's my sip.conf (well, the end of file) : http://pastebin.com/m7442f1ed |
| 10:50.18 | Coldfire_ | and my extensions.conf (at the end of [default]) : http://pastebin.com/m7c040946 |
| 10:50.35 | MaliutaLap | need the whole file, you're not telling us what context these things are dropping into |
| 10:51.05 | MaliutaLap | we need whole files |
| 10:51.13 | Coldfire_ | In fact I just add that to the default files |
| 10:52.32 | MaliutaLap | "default" means different things in different distribution/releases |
| 10:52.37 | MaliutaLap | the whole files |
| 10:53.03 | MaliutaLap | a) read the book. b) provide _all_ the information if you want help |
| 10:53.03 | iamy_china | Coldfire_: MaliutaLap is right |
| 10:53.13 | MaliutaLap | iamy_china: I normally am |
| 10:53.32 | iamy_china | MaliutaLap: :-) good for you |
| 10:53.50 | Coldfire_ | Ok, I'll read the book first. Thx |
| 10:55.24 | iamy_china | Coldfire_: good luck |
| 10:58.34 | joobie | boys anyoen played with polycom phones much? |
| 10:58.39 | joobie | trying to figure out how to turn on the LED |
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| 11:09.22 | nikpakar | hi anyone can help me to build a sip/ss7 gateway with asterisk and sangoma card |
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| 11:38.53 | shyam_k | whats the best way to record a dial() conversation? |
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| 11:41.11 | shyam_k | monitor()? |
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| 11:47.09 | shyam_k | hello?! |
| 11:47.29 | shyam_k | hi esaym am trying to monitor a call.. but it isn't recording the file.. |
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| 11:49.19 | tzafrir_laptop | shyam_k, it's generally not a good idea to ask specific people here |
| 11:49.34 | tzafrir_laptop | and yes, Monitor() is basically what you're looking for |
| 11:50.11 | esaym | Does the directory have the proper permissions? Running " asterisk -r " will take you to the asterisk console where you can see what is happening. in the asterisk console you can try increase the verbositiy with " core set verbose 9 " |
| 11:50.11 | tzafrir_laptop | Monitor sets the channel into recording mode |
| 11:50.47 | tzafrir_laptop | esaym, what do you mean by "proper permissions"? |
| 11:51.05 | tzafrir_laptop | permission for who to do what? |
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| 11:53.10 | joobie | omgg |
| 11:53.16 | shyam_k | tzafrir_laptop: yeah... sorry abt that.. and i got it working .. |
| 11:53.19 | joobie | how hard is it to find info on polycom led indicators |
| 11:53.27 | shyam_k | but the audio quality is very poor.. |
| 11:53.32 | joobie | anyone know how to configure these to trigger on certain events? |
| 11:53.53 | joobie | i found it in the phones config, where you can specify color, frequency, etc.. but duno how to trigger them from asterisk |
| 11:53.58 | esaym | tzafrir_laptop: permissions is basic unix stuff: http://en.wikipedia.org/wiki/File_system_permissions |
| 11:54.09 | shyam_k | any way to increase the audio quality? |
| 11:57.13 | esaym | shyam_k: use 711/ulaw codec |
| 11:59.40 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 12:02.35 | shyam_k | it seems something like specifying those pcm files in arecord.. |
| 12:03.09 | shyam_k | i mean the noice seem similar to that to a plain arecord which gets cured when we give some .asoundrc |
| 12:10.26 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:10.39 | shyam_k | where are the standard docs for these? a dialplan doc? describing automon, monitor and all? |
| 12:11.00 | [TK]D-Fender | shyam_k: ... |
| 12:11.02 | [TK]D-Fender | ~book |
| 12:11.02 | 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 |
| 12:11.14 | [TK]D-Fender | shyam_k: "cors show applications" , " core show application [appname]" |
| 12:11.32 | [TK]D-Fender | shyam_k: "core show functions" , " core show functions [funcname]" |
| 12:11.38 | [TK]D-Fender | core* |
| 12:11.54 | [TK]D-Fender | shyam_k: And the sample configs & WIKI |
| 12:11.55 | [TK]D-Fender | ~wikis |
| 12:11.56 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 12:12.19 | shyam_k | okay.. i have tfot.. |
| 12:14.18 | *** mode/#asterisk [+o [TK]D-Fender] by ChanServ |
| 12:16.00 | *** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender |
| 12:17.26 | joobie | TK |
| 12:17.28 | joobie | you alive? |
| 12:17.55 | joobie | I need your wisdom oh wise one |
| 12:18.47 | [TK]D-Fender | joobie: Shoot |
| 12:18.53 | *** join/#asterisk zxd (n=marceloa@84-16-228-34.internetserviceteam.com) |
| 12:18.54 | zxd | I noticed by default asterisk user is created with asterisk but with shell set to /bin/false , is there any security risks by changing it to /bin/bash , I want to run some scripts from asterisk |
| 12:19.05 | zxd | in debian lenny |
| 12:19.17 | joobie | two issues TK.. first (and hardest), any idea how to control the LED's on the polycom phones? specifically the IP 320 ? |
| 12:19.52 | joobie | the admin manual covers it briefly - saying you can configure the frequency, color, on/off state with the boot config, but it doesnt say how you can trigger it |
| 12:20.11 | [TK]D-Fender | zxd: there is no "default" and Asterisk does not create any users |
| 12:20.20 | joobie | been googling reading for the past hour.. not having much luck |
| 12:20.30 | zxd | [TK]D-Fender, I know In debian lenny |
| 12:20.48 | [TK]D-Fender | zxd: Go ask their packager then. |
| 12:21.09 | zxd | wouldnt you think it's unsafe running asterisk as root? |
| 12:21.19 | [TK]D-Fender | zxd: and the security risk is rather obvious. If someone can log as "asterisk" it can do what the user can. |
| 12:21.44 | [TK]D-Fender | zxd: And yes, it should be safer running * as another user than root |
| 12:22.01 | [TK]D-Fender | joobie: there is no such thing as "just controlling the lights" |
| 12:22.28 | [TK]D-Fender | joobie: The only thin you have any control over is presence on a speed-dial assigned line-key |
| 12:22.53 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 12:24.07 | joobie | TK, what is a speed-dial assigned line-key ? |
| 12:24.32 | joobie | well more specifically, which are the line-keys :P i have line1 and line2 keys? |
| 12:24.48 | joobie | but they are used when you grab a line |
| 12:25.06 | [TK]D-Fender | joobie: Look at it as you have control over *1* of those since the other is reserved for the minimal call-appearance you can have no it |
| 12:25.12 | *** join/#asterisk HenrikBe (n=zapphir@213.64.5.204) |
| 12:25.51 | [TK]D-Fender | joobie: so "Line1" can be for your reg, and the 2nd left empty so your contact directory can spill over and from there you enable "buddy watch" on it for presence. |
| 12:26.20 | *** join/#asterisk funkknob (n=funkknob@117.55.250.213) |
| 12:27.11 | joobie | i vaguely understand you |
| 12:27.28 | joobie | have you read the "Customizable Fonts and Idicators" section in the polycom admin guide? |
| 12:27.48 | joobie | .. it says "LED flashing sequences and colors can be changed" |
| 12:28.13 | [TK]D-Fender | joobie: that only changes how it flashes for events like ringing, hold, etc |
| 12:28.56 | joobie | TK, to give you the full picture.. i use AddQueueMember() via the dialplan to add a phone to the queue.. when it's in the queue, i want the phone to somehow show this visually |
| 12:28.57 | *** join/#asterisk [netman] (n=netman@51.Red-83-45-1.dynamicIP.rima-tde.net) |
| 12:29.06 | joobie | currently you join the queue and you wouldnt know the phone was apart of it |
| 12:29.14 | joobie | hence why i was thinking about using the LED |
| 12:29.23 | joobie | if it's not possible, any other ideas you can think of to do it? |
| 12:29.36 | joobie | i'm not too keen on using the LINE2 indicator as i wanted to keep this for a 2nd line on the phone |
| 12:33.46 | [TK]D-Fender | joobie: use the MicroBrowser Idle screen instead |
| 12:34.52 | joobie | TK, in english, is that the LCD display?:P |
| 12:35.00 | joobie | .. can i update the display via the dialplan? |
| 12:35.18 | joobie | like.. addqueuemember().. update microbrowser display()... |
| 12:35.38 | [TK]D-Fender | joobie: Go read the admin guide |
| 12:36.29 | *** join/#asterisk eliel (n=eliels@200.61.172.61) |
| 12:37.24 | funkknob | I have a problem with matching the context on an inbound call from a sip external peer. There are two peers configured identically, idd and vno but idd matches and correctly places the call into from-internal context, while vno always places the call into the from-trunk context. Both are identified by IP address in the host= statement and both use their IPs in the SIP headers. Both also have a seperate outbound peer config using from-tunk context ( |
| 12:37.25 | funkknob | since this is a Trixbox). I'm using host=<IP>, insecure=port and type=friend. How can I determine what is preventing a match? |
| 12:37.49 | joobie | TK, ok.. but just as a quick yes / no (so i can get my hopes up).. is it possible to update the microbrowser from the dialplan? |
| 12:38.03 | tzafrir_laptop | zxd, where does that shell bother you? |
| 12:38.13 | [TK]D-Fender | joobie: That is not how it updates. |
| 12:40.25 | joobie | ahh k |
| 12:40.34 | joobie | i'll read the manual |
| 12:40.38 | joobie | thanks for the heads up |
| 12:40.48 | joobie | the other issue TK, probably a bit more simpler to resolve |
| 12:40.50 | joobie | btu im stumped. |
| 12:42.08 | joobie | i've setup a queue which i add dynamic members to.. when i put the members in the queue on pause, if a new call comes in, queue() doesn't accept the new caller into the queue (only if all handsets are on pause or if no one is dynamically added as a member)... i set the variabels 'joinempty=yes' and 'leavewhenempty=yes' but didn't resolve it |
| 12:42.30 | joobie | supposedly joinempty=yes should fix it and allow users to join empty queues.. |
| 12:44.39 | *** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net) |
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| 12:56.12 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-182-243-rbry-bb-1.dynamic.isadsl.co.za) |
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| 13:05.02 | freh | joobie, try leavewhenempty=no |
| 13:06.32 | joobie | bingo |
| 13:06.33 | joobie | that worked |
| 13:06.51 | joobie | i thought that should be set to yes? |
| 13:07.10 | joobie | i guess when you read the variable, it's pretty straight forward |
| 13:07.20 | joobie | .. relative to the caller |
| 13:07.21 | freh | indeed :p |
| 13:07.23 | joobie | heh |
| 13:07.25 | joobie | thanks freh |
| 13:07.29 | freh | np |
| 13:07.51 | joobie | freh have you played with that periodic-announcement setting? |
| 13:08.09 | joobie | err periodic-announce setting even |
| 13:08.20 | freh | yes |
| 13:08.58 | joobie | hmm.. trying to achieve ideally.. a queue that after the first 20 seconds, plays wavfile_1.. then after that, it plays wavfile_2 every 30 seconds |
| 13:09.11 | *** join/#asterisk Gabriel25 (n=gabe@pool-72-68-157-205.nycmny.fios.verizon.net) |
| 13:09.22 | joobie | so kinda have two periodic announcements.. the first one.. then a different one there after.. and with different freuqencies |
| 13:09.24 | Gabriel25 | hi guys ... |
| 13:09.27 | joobie | do you know if that's possible? |
| 13:09.35 | Gabriel25 | somene heard about www.genxvoip.com ?? |
| 13:09.59 | [TK]D-Fender | joobie: vi app_queue.so |
| 13:10.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:10.04 | [TK]D-Fender | joobie: vi app_queue.c rather |
| 13:10.30 | joobie | ahh serious |
| 13:10.58 | freh | don't know. I just have standard music playing and every 60 seconds periodic announce |
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| 13:11.25 | joobie | hmmm |
| 13:11.30 | joobie | maybe i can setup two queues |
| 13:11.39 | joobie | and bounce the user to another queue after 20s |
| 13:11.58 | joobie | so sorta drop the user into queue1.. which has the 20s announcement |
| 13:12.08 | joobie | then drop them to queue2, which has the 30s, with the different wave |
| 13:12.15 | joobie | or does that sound gay...... |
| 13:13.30 | [TK]D-Fender | joobie: Will be harder to collect queue stats, etc, plus the limited ring time |
| 13:13.40 | [TK]D-Fender | joobie: Pretty gay :p |
| 13:13.45 | joobie | true |
| 13:13.50 | joobie | didn't consider queue stats |
| 13:13.51 | joobie | argh |
| 13:14.32 | freh | joobie, just play wavfile_1 with Playback() before the join the queue |
| 13:14.46 | joobie | these asses are being difficult |
| 13:14.57 | joobie | they want a "oh hi there.. ur on hold waiting.. it wont be long" |
| 13:14.57 | freh | and use wavfile_2 as periodic-announce for the queue |
| 13:15.07 | joobie | then after that they want the "oh.. it's been a while, leave a message" |
| 13:15.27 | joobie | so gotta really queue the caller, wait 20s then play the first wave |
| 13:18.35 | [TK]D-Fender | joobie: There is *1* option for this. If MoH can be set so it always starts from the beginning you could make a single sound file with the complete "on-hold" music mixed with your messages. |
| 13:18.52 | freh | you could just put the message in the music on hold files |
| 13:18.52 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.68.173) |
| 13:19.44 | joobie | TK |
| 13:19.52 | joobie | you are a fuken genius |
| 13:20.06 | joobie | you too freh ;P |
| 13:20.13 | joobie | that will work a charm |
| 13:21.34 | freh | remember that the music file will loop. So if they're on hold long enough they'll hear the first message again |
| 13:21.48 | joobie | ahh |
| 13:21.49 | joobie | ahhhhh |
| 13:21.53 | joobie | yea that could be a problem. |
| 13:21.55 | *** join/#asterisk xrmx__ (n=rm@host128-22-dynamic.15-87-r.retail.telecomitalia.it) |
| 13:22.08 | joobie | the music file plays only once for the whole queue ya? |
| 13:22.27 | joobie | so if a caller joins.. it starts playing.. if another joins 10 seconds into it, they will not hear it from the start ya? |
| 13:22.48 | joobie | i think i read this.. something about less load on the asterisk box having the one stream for the queue instead of a seperate one for each caller |
| 13:23.22 | freh | not sure about that. In fact that's something I'd like to know too |
| 13:25.02 | joobie | sec i might have reference to it |
| 13:25.14 | joobie | but ya.. that shoots that idea down :/ |
| 13:26.52 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 13:27.03 | joobie | http://www.orderlyq.com/asteriskqueues.html |
| 13:27.07 | joobie | Note: If you're using Asterisk 1.2.x or 1.4.x with mode=files, each new caller will hear your on-hold music from the start (rather than where it left off from the previous call as with Asterisk 1.0.x). If this behaviour is not desired, you can use the rawplayer as described above, or compile a rawplayer from the source in /usr/src/asterisk-1.2.x/contrib/utils - see the README in that directory for more information. |
| 13:27.21 | joobie | my bad.. there's an ption to force it to play from the start |
| 13:28.30 | joobie | ahh actually that's saying about the stop point of the wave/mp3 |
| 13:28.47 | joobie | like say the last caller stops at 00:56.. does it start at 00:57 or back at 00:00 |
| 13:33.00 | *** join/#asterisk dror99 (i=d4b38cc2@gateway/web/ajax/mibbit.com/x-d1837e906812eb33) |
| 13:33.34 | dror99 | hi |
| 13:34.12 | dror99 | ~take-a-number |
| 13:34.20 | russellb | blinks |
| 13:34.20 | anonymouz666 | lol |
| 13:34.29 | anonymouz666 | this is not freeswitch |
| 13:34.30 | anonymouz666 | heh |
| 13:35.37 | *** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net) |
| 13:35.38 | dror99 | I have a question. We get the following error "app_dial.c: Could not stop autoservice on calling channel", what does it mean? |
| 13:37.35 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 13:40.33 | joobie | need sleep |
| 13:40.34 | joobie | nite |
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| 14:01.24 | *** join/#asterisk Holos (n=cosmond@209.167.131.35) |
| 14:01.55 | Holos | Hey, anyone had any experience with the Rowtel IP01 / IP04 (Blackfin Asterisk with 1 or 4 FXO/FXS) |
| 14:02.15 | Holos | Flash drive, no moving parts, OSLEC, and FXO's make it sound nice. |
| 14:04.31 | dror99 | ~take-a-number |
| 14:04.31 | infobot | 17 |
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| 14:07.39 | dror99 | Can you please explain is short what is autoservice on a channel? |
| 14:08.59 | tzafrir_laptop | what version of Asterisk do you use? |
| 14:09.51 | *** join/#asterisk awk_r (n=awk_r@nat/digium/x-58c4b32690673152) |
| 14:10.11 | dror99 | 1.4.21.2 |
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| 14:12.46 | tzafrir_laptop | well, there was an autoservice and removing is has failed. I also see that the return value from this function is only checked in two of the four places in which it is called |
| 14:12.56 | tzafrir_laptop | That's as far as I can see |
| 14:13.01 | tzafrir_laptop | anybody else? |
| 14:14.31 | dror99 | What is autoservice on a channel? |
| 14:15.10 | russellb | it's not something that really makes sense from the user perspective ... |
| 14:15.15 | russellb | it's an internal channel handling detail |
| 14:15.35 | *** join/#asterisk pwebguy (n=pwebguy@200.110.240.130) |
| 14:15.49 | russellb | However, it really should _never_ fail |
| 14:19.49 | [TK]D-Fender | dror99: perhaps you should upgrade as well. |
| 14:22.00 | carrar | upgrades his iPhone |
| 14:25.37 | *** join/#asterisk Dovid (n=annon@ool-4355e297.dyn.optonline.net) |
| 14:26.00 | Dovid | hi, |
| 14:26.05 | carrar | Hi!! |
| 14:26.20 | carrar | Ohayoo gozaimasu! |
| 14:26.41 | Dovid | anyone have an issue where asterisk is behind NAT. sip telephones locally have no issue. if i try to connect remotely i get an error from asterisk that the password is not valid |
| 14:27.15 | Dovid | if i try the same account locally there is no issue. also if i delete the password then i can register remotely (aka with nat) with no issue. never seen such a thing before |
| 14:28.52 | *** join/#asterisk JayTee52 (n=jforde@unaffiliated/jaytee) |
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| 14:36.04 | eppigy | hello |
| 14:40.30 | [TK]D-Fender | eppigy: you are dave |
| 14:43.28 | jbjuly | I'm trying to setup a MeetMe conference, with an admin flag, but when the admin user joins the conference first, the normal users will get a "invalid pincode", is there any workaround on how to make the conference joinable when an admin user is joined? |
| 14:43.48 | freh | Is there some application in which you can gather asterisk call statistics? |
| 14:44.20 | dror99 | russellb: What is you suggestion regrading this issue? Should we do something regarding this error? Should we ignore it? We get it 4 times a day... |
| 14:45.53 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 14:45.53 | *** mode/#asterisk [+o denon] by ChanServ |
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| 14:47.50 | russellb | dror99: It's up to you. It's probably a bug. If it's causing you problems, then please report it. If not, then it's up to you whether you want to report it and help us test/debug it. |
| 14:48.52 | Dovid | TK: see my question |
| 14:48.53 | Dovid | ? |
| 14:53.26 | *** join/#asterisk C4colo (n=DJpyro@66.185.111.33) |
| 14:54.28 | C4colo | if I have all of my extensions and all of my trunks using g729, all calls should require no transcoding right? ... but how does this work in conferences that use slin for the codec inside of meetme? |
| 14:54.59 | [TK]D-Fender | C4colo: MeetMe Will transcode everything and that'll be a big hit |
| 14:55.23 | C4colo | that is what I thought |
| 14:55.55 | C4colo | so I need 0 licenses if 100% is g729, but I need 10 licenses to have 10 people in one conference |
| 14:56.00 | *** join/#asterisk n3hxs (n=HAMming@63.68.135.4) |
| 14:56.06 | C4colo | hmm |
| 14:56.06 | [TK]D-Fender | C4colo: Also you'll have to tweak every recording app to do so in G.729 (VM, etc) and make sure you have matching prompts, etc |
| 14:56.21 | C4colo | yea, I'll get a few channels anyway just to be safe |
| 14:56.40 | C4colo | but I didn't want to have to buy a ton just to use conferencing |
| 14:56.50 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:57.06 | C4colo | I guess I'll get a few and just put the remote extensions on g729, leave everything else ulaw |
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| 14:58.07 | dror99 | thanks :-) |
| 14:58.29 | *** part/#asterisk dror99 (i=d4b38cc2@gateway/web/ajax/mibbit.com/x-d1837e906812eb33) |
| 15:01.59 | C4colo | thanks [TK]D-Fender |
| 15:02.13 | *** join/#asterisk zapotek6 (n=edpman@mail.comelit.it) |
| 15:02.20 | [TK]D-Fender | C4colo: Watch out of mixmonitor as well |
| 15:05.00 | C4colo | if I have 10 extensions and only two of them need g729, and I only enable that codec on those two extensions, it should only transcode once regardless of how many applications are on that channel right? |
| 15:05.26 | C4colo | so I would need two licenses if only the leg that is going out to those extensions is g729 |
| 15:05.46 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 15:07.07 | [TK]D-Fender | C4colo: ? |
| 15:07.56 | C4colo | regardless of if that channel is using mixmonitor and meetme, if it is just g729 out to that phone I only need one license for that one extension right? |
| 15:08.08 | C4colo | or could mixmonitor use one license plus meetme use another license? |
| 15:08.16 | C4colo | theoretically, I can't think of a good example other than that |
| 15:10.45 | Kobaz | anyone know how i can get an audiocodes gateway to send inband dtmf (so that i can use stuff defined in features.conf) |
| 15:11.14 | [TK]D-Fender | C4colo: Includ chanspy, etc in that equaion. the math is simple. Every transcode takes 1 |
| 15:11.21 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 15:11.44 | [TK]D-Fender | Kobaz: AC should send RFC2833, and you shouldn't be looking for inband |
| 15:12.04 | Kobaz | [TK]D-Fender: it is rfc2833 |
| 15:12.13 | Kobaz | i'm not seeing any sip packets for dtmf |
| 15:12.50 | Kobaz | i tried setting notify and info too |
| 15:12.51 | Kobaz | nothing |
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| 15:16.50 | *** join/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com) |
| 15:17.55 | _brent_ | `asterisk -vvvc` should start asterisk and give me a console, shouldn't it? |
| 15:18.04 | *** join/#asterisk af_ (n=getsmart@88-149-240-185.dynamic.ngi.it) |
| 15:18.12 | _brent_ | i'm running 1.6.1 rc5 and it starts, but returns |
| 15:18.25 | Kobaz | [TK]D-Fender: anything else i should check? |
| 15:18.45 | _brent_ | i can then connect with `asterisk -r` but asterisk forking and returning breaks safe_asterisk |
| 15:19.40 | [TK]D-Fender | Kobaz: I'm not ocmmenting blind. |
| 15:20.26 | _brent_ | hmpf. alwaysfork was set. |
| 15:21.38 | _brent_ | for the sake of posterity, safe_asterisk doesn't work if alwaysfork=yes in asterisk.conf |
| 15:22.46 | *** join/#asterisk wpbrown (n=wpbrown@206.251.162.2) |
| 15:41.01 | *** join/#asterisk proxium (n=proxium@196.203.51.238) |
| 15:41.22 | proxium | Hello, how to resolv such error ?Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/50000000-081f9648' in macro 'hangupcall' |
| 15:42.20 | proxium | this happen when I call myself, or logged into vicidial |
| 15:42.26 | Kobaz | proxium: don't hang up? |
| 15:42.35 | _brent_ | proxium: that's not an error, it just hung up |
| 15:42.36 | Kobaz | proxium: and that |
| 15:42.40 | Kobaz | 's not an error |
| 15:42.58 | _brent_ | judging by your macro's name, it sounds like that's what is desired? |
| 15:43.15 | proxium | I accept the comunication but it stop ringing and ws interrupted |
| 15:44.32 | pwebguy | Hullo all. I am beating my head against the wall with the IAX2 vice encryption feature. I have foollowed the instructions on voip-info.org as well as a few other sites that I have found, and no matter the encryption settings I am always able to play back my conversations after capturing on the local machine or the server. Is this still a 'feature', or was it dropped? Anyone have any experience with this? |
| 15:45.01 | _brent_ | proxium: in the CLI, type `core set verbose 9` and you'll see more output. that may help you see what the dial plan is doing. |
| 15:45.29 | proxium | _brent_: ok I'll do |
| 15:45.57 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:46.11 | russellb | pwebguy: what version? |
| 15:46.22 | russellb | and how are you doing the capture? |
| 15:46.49 | russellb | and are you calling between two asterisk boxes? or another client? |
| 15:48.43 | pwebguy | version 1.6 (the tar from currentversion, re-downloaded a couple days ago) |
| 15:48.55 | pwebguy | the capture is happening with tcpdump on the server and |
| 15:49.09 | pwebguy | with wireshark and unsniff on my local pc |
| 15:49.22 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 15:49.24 | Faustov | hi |
| 15:49.32 | pwebguy | These are calls between clients: client - asterisk - client |
| 15:49.38 | russellb | what is the client? |
| 15:49.56 | pwebguy | I am testing with zoiper, and also have tested with kiax |
| 15:50.00 | ltd | Is it possible to have the sip_pvt->callingpres value somehow carry over to a bridged/dialled zap channel? |
| 15:50.09 | Faustov | to use a specific codec, it has to be enabled on both sides, right? As in, if some party enforces codec A, I'll have to end up using it? |
| 15:50.11 | russellb | pwebguy: are you sure those clients support IAX2 encryption? |
| 15:50.15 | russellb | I don't think they do. |
| 15:50.56 | pwebguy | I could not find any documentation stating one way or the other; however to cover this I also added forceencryption=yes to the iax.conf |
| 15:51.27 | pwebguy | What clients support it for sure? (google is not helping much with this?) |
| 15:51.28 | russellb | that is not supported in 1.6.0 |
| 15:51.37 | russellb | that is a new feature only in trunk (and maybe 1.6.2) |
| 15:51.41 | russellb | I don't know of _any_ client that supports it |
| 15:51.46 | russellb | other than asterisk to asterisk .. |
| 15:51.50 | pwebguy | hang on, I will get you the exact version |
| 15:52.27 | pwebguy | Asterisk 1.6.0.9 |
| 15:52.31 | proxium | _brent_: this is the output when I call a number: -- AGI Script dialparties.agi completed, returning 0 |
| 15:52.31 | proxium | <PROTECTED> |
| 15:52.31 | proxium | <PROTECTED> |
| 15:52.32 | proxium | <PROTECTED> |
| 15:52.32 | proxium | <PROTECTED> |
| 15:52.32 | proxium | <PROTECTED> |
| 15:52.34 | proxium | <PROTECTED> |
| 15:52.38 | Faustov | ffs |
| 15:52.49 | Kobaz | proxium: pastebin |
| 15:52.58 | proxium | ok sorry |
| 15:53.03 | russellb | pwebguy: Okay, so, forceencryption is not ssupported in 1.6.0. |
| 15:53.09 | russellb | Also, I don't think those clients support encryption. |
| 15:53.12 | russellb | So, that explains what you see. |
| 15:53.45 | pwebguy | I'll check out 1.6.2. Yes, that explains a lot! Do you know of any softphone clients that support this? |
| 15:53.58 | pwebguy | OR possibly an ATA or hardphone? |
| 15:54.17 | russellb | I recall Tim Panton from phonefromhere.com saying that he had implemented it in his stack |
| 15:54.20 | russellb | but that's all I know of |
| 15:54.30 | Faustov | could anyone help me with the codec question above please? |
| 15:54.39 | [TK]D-Fender | pwebguy: Extremely few IAX phones out there, all suck and are old standard... |
| 15:54.52 | pwebguy | Ok, thanks a lot russelB, I really appreciate it |
| 15:54.56 | russellb | pwebguy: you're welcome |
| 15:55.02 | pwebguy | D-Fender - yes, that is what I am learning |
| 15:55.18 | [TK]D-Fender | Faustov: What are "both sides"? |
| 15:55.47 | proxium | this the complete output when I manually dial 003355559999 http://pastebin.com/m54156a28 |
| 15:55.54 | pwebguy | Our operation requires voice encryption, so I am going to explore this IAX a bit more, then go back to playing with the SRTP/TLS |
| 15:55.55 | Faustov | [TK]D-Fender: side one -> asterisk <- iax2 provider <- other side |
| 15:56.27 | [TK]D-Fender | Faustov: No, both sides do not need to speak the same codec. Thats what TRANSCODING is for |
| 15:56.54 | [TK]D-Fender | proxium: Enable SIP debug to see whats actually going on. You're only looking at part of the picture there |
| 15:57.16 | proxium | :) ok |
| 15:57.54 | kc8pxy | [TK]D-Fender: is there any way to specify where the transcoding takes place? |
| 15:58.06 | Kobaz | pwebguy: openvpn |
| 15:58.22 | kc8pxy | [TK]D-Fender: as in, which box/phone does the transcoding? |
| 15:58.22 | Faustov | [TK]D-Fender: nice, how is transcoding managed/configured in asterisk? |
| 15:58.54 | ValDuane | anyone know a good place to start looking for a freelancer for a small asterisk aplication project? |
| 15:59.01 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:59.31 | [TK]D-Fender | kc8pxy: DEVICES don't transcode, ASTERISK does |
| 16:01.00 | proxium | [TK]D-Fender: http://pastebin.com/m6839a1f9 this is done with SIP Debug and core Debug level 9 |
| 16:01.16 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 16:01.21 | Faustov | [TK]D-Fender: could you please point me to documentation where managing transcoding is explained? |
| 16:01.52 | SuPrSluG | proxium: r u using trixbox with vicidial? |
| 16:02.37 | pwebguy | Kobaz: Yes, looked into OpenVPN also, that may be the final option but I am trying to keep everything as simple as possible. |
| 16:03.18 | Kobaz | pwebguy: openvpn is really simple |
| 16:03.30 | *** join/#asterisk bijit (n=benji@190.241.15.48) |
| 16:03.52 | Kobaz | pwebguy: generate ca cert, server cert, client cert, edit the default config... connect... done |
| 16:04.29 | Kobaz | apt-cache show imagemagick |
| 16:04.30 | pwebguy | Yes, certainly I have set it up for my own use many times - But I am dealing with office workers who are not technically savvy so I am trying to make it as close to 'point and click' as possible. |
| 16:04.31 | Kobaz | er |
| 16:04.35 | proxium | SuPrSluG: no I'm using Asterisk (Ver. 1.4.22) ==> Asterisk (Ver. 1.2.136945) (a local trunk) and FreePBX and Vicidial 2.0.5 in a separate machine |
| 16:05.21 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 16:05.50 | SuPrSluG | proxium: you may be missing the 'o' option in the dialstring. check here http://iptn.org/vicidial/index.html |
| 16:07.15 | pwebguy | Kobaz: I am saving OpenVPN for the final option. IAX2 (from what I could find on the web) was a bit of a panecia (all in one, no nat issues, etc) so I was really hoping it would work. The work on the SRTP branch looks promising, zoiper biz supports it (although I would MUCH rather find an opensource client). So, we'll see what happens. OpenVPN may be the solution |
| 16:07.32 | Kobaz | open source iax clients suck ass |
| 16:07.35 | Kobaz | sadly |
| 16:08.08 | pwebguy | yes, calls across the pond didn't work too well, but local calls were not bad. |
| 16:08.14 | Faustov | kc8pxy: do you know anything about transcoding in asterisk? |
| 16:08.20 | pwebguy | It is useless to me without the voice encryption though |
| 16:08.34 | *** join/#asterisk acxty (n=acxty@201.220.136.117) |
| 16:09.15 | pwebguy | Anyone know of any open-source clients that support SRTP/TLS? |
| 16:10.11 | *** join/#asterisk astassistant (n=skip@65-126-63-1.dia.static.qwest.net) |
| 16:11.41 | *** join/#asterisk MrNaz (n=mrnaz@ppp118-208-245-125.lns10.mel6.internode.on.net) |
| 16:12.01 | proxium | SuPrSluG: I use FreePBX to do the most of Job and I'm newbie in Asterix, In "Asterisk Dial command options:" I have rTtr and I don't know where to insert 'o' ! |
| 16:12.56 | SuPrSluG | proxium: replace the second r with o . |
| 16:13.23 | acxty | Hi guys, may someone guide me a little with this. What I want to do is to capture some data from a database. I will use agi with php + mysql. That part I know what to do. What I am not sure is the next part. After I have that information I want it to pass it to asterisk and make a menu with it. For example press 1 for resutl1, press 2 for result2, press n for resuln |
| 16:13.25 | SuPrSluG | proxium: you don't need it it's already there |
| 16:13.33 | proxium | SuPrSluG:on my outbound server ? |
| 16:13.40 | SuPrSluG | proxium: yes |
| 16:13.43 | *** part/#asterisk _brent_ (n=_brent_@166-70-142-225.ip.xmission.com) |
| 16:14.07 | beek | acxty: Lookup up func_odbc |
| 16:14.15 | acxty | thanks |
| 16:15.29 | SuPrSluG | proxium: whatever is doing the outbound calls |
| 16:20.01 | *** join/#asterisk juanIMP (n=Juancho@190.26.210.241) |
| 16:21.39 | *** join/#asterisk proxium (n=proxium@196.203.51.238) |
| 16:22.08 | proxium | SuPrSluG: the phone hangup automatically |
| 16:23.35 | *** join/#asterisk ice_croft (n=nolan@85.172.5.106) |
| 16:26.24 | proxium | I receive a message from Vicidial: Customer has hung up: SIP/tofreepbx1-09a20ad8 |
| 16:29.04 | *** join/#asterisk taylorS (n=taylorS@173.9.54.69) |
| 16:30.16 | taylorS | help |
| 16:31.42 | taylorS | is anyone here? |
| 16:33.00 | tzafrir_laptop | no |
| 16:33.11 | tzafrir_laptop | ~ask |
| 16:33.11 | 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. |
| 16:33.44 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 16:34.37 | taylorS | eye eye: i'm a newbie working with someone who has setup a functional askterisk system. i have 2 phones that are not able to use dhcp to connect to the boot server. i'm looking for any troubleshooting ideas. |
| 16:35.04 | tzafrir_laptop | what phones? |
| 16:35.23 | SuPrSluG | put the ip address in |
| 16:35.25 | taylorS | the phones cannot find the server and keep rebooting. unfortunately, i don't have the server password. however, since the rest of the phones in the office work (over the same switch), i'm assuming it is not a network issue |
| 16:35.26 | tzafrir_laptop | anyway, a sniffer can always help |
| 16:35.43 | [TK]D-Fender | Faustov: What don't you get? transcoding is where 2 endpoints each go through some central server and translates codec's, formatrs, etc |
| 16:35.53 | taylorS | polycom soundpoint ip |
| 16:36.20 | taylorS | the ip address is correct |
| 16:36.26 | Corydon76-dig | taylorS: I assume you've done the basic tests of ensuring wire continuity to the switch? |
| 16:36.27 | taylorS | how do i get a sniffer? |
| 16:36.32 | taylorS | yes |
| 16:37.03 | Corydon76-dig | taylorS: have you tried different ports on the switch? |
| 16:37.10 | [TK]D-Fender | taylorS: I'd confirm what IP its provisioning from and I'd get those password if I were you |
| 16:37.18 | taylorS | Corydon76-dig: yes, I've tried several different ports |
| 16:37.21 | *** join/#asterisk propellerhead (n=yogurt2u@host34.190-136-236.telecom.net.ar) |
| 16:37.34 | Corydon76-dig | taylorS: which DHCP server are you running? |
| 16:37.43 | Corydon76-dig | Windows? |
| 16:37.44 | seb- | [TK]D-Fender: I installed dahdi and * from source, dahdi drivers don't load automatically and when i try "modprobe dahdi_dummy" i get a segfault!?!? |
| 16:37.52 | taylorS | Corydon76-dig: computers are able to connect from those ports, but phones are not |
| 16:37.58 | [TK]D-Fender | seb-: Still running in a vM? |
| 16:38.04 | seb- | [TK]D-Fender: yes |
| 16:38.12 | [TK]D-Fender | seb-: Can't help you there |
| 16:38.37 | seb- | [TK]D-Fender: are you saying i should not run * in a VM? |
| 16:38.42 | Corydon76-dig | taylorS: Computers that were already on the network or computers that were not? |
| 16:38.50 | [TK]D-Fender | seb-: that's always been by opinion |
| 16:39.03 | seb- | [TK]D-Fender: do you think that might stop the segfaulting? |
| 16:39.07 | Corydon76-dig | taylorS: I'm thinking you're out of addresses in the DHCP pool |
| 16:39.19 | [TK]D-Fender | taylorS: go into your phone's bootrom and ocnfirm the server IP it has listed |
| 16:39.57 | *** join/#asterisk hfb (n=hfb@pool-96-247-49-46.lsanca.dsl-w.verizon.net) |
| 16:40.08 | taylorS | Corydon76-dig: definitely not out of addresses. am able to add and remove machines at multiple ports. Just the phones are not working. I am totally new to asterisk. Is there a good starter tutorial or quick troubleshooting guide? |
| 16:40.10 | [TK]D-Fender | Corydon76-dig: Could be improperly set phones (cyclical rebooting is usualy the side effect of corrupted configs). Esp as Polycom's don't need a boot server to start. they jsut use the last loaded settings upon failure. |
| 16:40.48 | taylorS | D-Fender: corrupted config is quite possible, am getting such an error message when the phone reboots |
| 16:40.53 | Corydon76-dig | [TK]D-Fender: yes, but they also need a valid IP to start |
| 16:41.01 | taylorS | D-Fender: if you give me a minute I can get the exact msg |
| 16:41.08 | [TK]D-Fender | Corydon76-dig: Yeah, failing to pick up an IP at all would be bad. |
| 16:41.22 | [TK]D-Fender | taylorS: Again go prove it failes to get even a basic IP |
| 16:42.59 | taylorS | Corydon76-dig: "failed to get boot parameters via DHCP"...the config error comes later |
| 16:43.49 | taylorS | Corydon76-dig: does that mean the phone is not able to assign itself an IP b/c did not find DHCP server? |
| 16:43.51 | [TK]D-Fender | taylorS: That doesn't mean failure to get an IP. What happens following? |
| 16:43.56 | SuPrSluG | taylorS:that's normal. the phone wants to contact an ftp/tftp server to get it's config |
| 16:44.05 | taylorS | D-Fender: updating initial config |
| 16:44.12 | [TK]D-Fender | taylorS: and at the end? |
| 16:44.38 | taylorS | D-Fender: waiting now...give you exact msg when done |
| 16:44.59 | tzafrir_laptop | seb-, get the patch from http://bugs.digium.com/view.php?id=13930 |
| 16:45.07 | SuPrSluG | taylorS:when it finishes get the ip address via the menu button and go to that ip in your browser |
| 16:45.09 | tzafrir_laptop | Hopefully it will be merged soon |
| 16:46.19 | SuPrSluG | taylorS:you only need to configure the Line tab for username,passwd and proxy |
| 16:46.55 | [TK]D-Fender | SuPrSluG: for the minimum. |
| 16:47.10 | *** join/#asterisk dr_gogeta86 (n=fisgro@81-208-88-100.ip.fastwebnet.it) |
| 16:47.13 | SuPrSluG | reference to wiki? |
| 16:47.24 | [TK]D-Fender | then again people configuring Polycom's directly on the phone UI or web UI should be dragged out and shot. Survivors should be shot AGAIN :p |
| 16:47.24 | dr_gogeta86 | hi to all |
| 16:47.42 | dr_gogeta86 | anyone come from italy i have many questions for you |
| 16:49.54 | taylorS | D-Fender: "error loading 0004f200edff.cfg" |
| 16:50.25 | taylorS | SuPrSluG: just downloaded the book and checked out the troubleshooting guide |
| 16:51.28 | jblack | omg. I just heard how bad calls to my voipstreet number is. It's _awful_ |
| 16:51.32 | [TK]D-Fender | taylorS: So... why don't you have passwords to your own system? |
| 16:51.52 | Faustov | [TK]D-Fender: http://forums.digium.com/viewtopic.php?t=68538 |
| 16:52.27 | taylorS | D-Fender: guy who configures is out and some of the office staff now can't use phones |
| 16:52.29 | SuPrSluG | taylorS: you can reset to default config by pressing 468* (most models) or 1357 (for 330's) |
| 16:52.56 | *** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 16:53.13 | taylorS | SuPrSlu |
| 16:53.20 | taylorS | G: 468*? |
| 16:53.35 | taylorS | SuPrSluG: at any point during config? |
| 16:54.06 | SuPrSluG | do it when the phone starts to boot |
| 16:54.34 | [TK]D-Fender | Faustov: NOTHING TO MANAGE! |
| 16:54.45 | [TK]D-Fender | Faustov: That is a very sad and broken post |
| 16:55.41 | [TK]D-Fender | Faustov: Configure a softphone for GSM only. Configure another SIP phone for G.711u only. Call through * and it with JUST HAPPEN. there is no "configuring! |
| 17:09.33 | SuPrSluG | taylorS:press and hold down all keys at once until asked for a password = 456 by default. |
| 17:09.55 | taylorS | SuPrSluG: thanks for the tip! fixed the settings according to other phones in the office and trying again now |
| 17:13.00 | *** join/#asterisk stevetotaro (n=Steve@c-69-243-124-5.hsd1.md.comcast.net) |
| 17:15.35 | Faustov | [TK]D-Fender: so how come i had only alaw and ulaw-only dialers couldn't connect? |
| 17:16.11 | Faustov | had as in, disallow=all, allow=alaw |
| 17:16.27 | [TK]D-Fender | Faustov: Because that is only ONE leg of the call |
| 17:16.59 | [TK]D-Fender | Faustov: What you allow IN from your dialer device has NOTHING to do with a DIAL that it will issue to call to another resource |
| 17:19.02 | *** join/#asterisk hi365 (n=hi365@94.159.178.61) |
| 17:22.56 | Faustov | [TK]D-Fender: so even if I transcode the incoming calls into something else I won't earn anything quality/bandwidth wise? |
| 17:24.42 | [TK]D-Fender | <PROTECTED> |
| 17:30.16 | Faustov | [TK]D-Fender: doesn't ANY codec try to recreate the non-discrete signal at all? That's where I'm trying to gain on quality... |
| 17:30.49 | seb- | [TK]D-Fender: the *only* reason i set up * is for meetme and that appears to be the *only* thing that has problems with Xen VMs!?!? :) |
| 17:30.57 | seb- | ahhhhh! |
| 17:31.10 | [TK]D-Fender | Faustov: You can't make crap BETTER. How do you restore an Mp3 to CD quality? You CAN'T its against the laws of physics. transcoding is like phtocopying, you always lose in the translation. |
| 17:31.22 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 17:31.27 | [TK]D-Fender | seb-: Life sucks, but rarely swallows :) |
| 17:31.42 | seanbright | i'll be here all week |
| 17:31.45 | seanbright | try the veal |
| 17:33.20 | [TK]D-Fender | is wondering where so many people are getting their crack from.... |
| 17:33.45 | Faustov | [TK]D-Fender: yeah but in plain PCM you send 64kbps for the sound stream, and problems start happening when for example one packet gets dropped. ADPCM codecs however use less bandwidth because there is prediction, which lets the codec re-create the sound. Recreated sound is obviously lower quality, because there is prediction error, but that is still better than no sound probe for this fraction |
| 17:33.48 | Faustov | of second at all |
| 17:33.52 | seb- | [TK]D-Fender: crack.com ? |
| 17:34.31 | [TK]D-Fender | Faustov: Now you're talking PACKET LOSS? Seriously... pick a track and stick with it |
| 17:35.21 | Faustov | [TK]D-Fender: that was just an example, a probe in a sound stream can be dropped due to various reasons: delay, packet loss, bad frame, whatever |
| 17:35.23 | [TK]D-Fender | Faustov: lost of a same-time-sized packet should be the same scale of quality loss. |
| 17:35.42 | [TK]D-Fender | Faustov: And this has abosultely notihng to do with transcoding. |
| 17:35.55 | Faustov | i agree it has nothing to do with transcoding |
| 17:36.07 | Faustov | my intention is to find out if transcoding can be a workaround for this |
| 17:37.01 | [TK]D-Fender | Faustov: No. PLC is what occurs between endpoints, and * IS an endpoint. |
| 17:37.17 | Faustov | k, too bad |
| 17:37.49 | [TK]D-Fender | Faustov: This happens, THEN transcoding happens, THEN it passes the packet on to another leg where MORE PL can occur and more degradation. |
| 17:38.11 | Faustov | ok, that makes sense |
| 17:38.26 | [TK]D-Fender | Faustov: If * is on the lossy side prepare to get screwed both ways |
| 17:38.30 | Faustov | by the way, g729 has transcoding licenced, right? |
| 17:38.37 | [TK]D-Fender | Faustov: Yes |
| 17:38.43 | Faustov | ok |
| 17:41.58 | seb- | [TK]D-Fender: tell your friends! i found 2 possible alternatives to meetme/ztdummy by googling! |
| 17:42.28 | seb- | [TK]D-Fender: 1. use app_conference instead of meetme or 2. use ztxen instead of ztdummy |
| 17:42.33 | seb- | [TK]D-Fender: haven't tested yet |
| 17:42.42 | seb- | [TK]D-Fender: http://blogs.osuosl.org/gchaix/2006/07/17/asterisk-and-xen/comment-page-1/ |
| 17:43.00 | dr_gogeta86 | hi to all |
| 17:43.13 | dr_gogeta86 | who can help me with fxo and tdm400 |
| 17:43.57 | [TK]D-Fender | dr_gogeta86: Indications problem? |
| 17:44.25 | dr_gogeta86 | i have succesfull configured fxs ports |
| 17:45.00 | dr_gogeta86 | but i can't understand how to connect pstn line to fxo port and how to configure zapata and asterisk for incoming call |
| 17:45.00 | *** part/#asterisk pwebguy (n=pwebguy@200.110.240.130) |
| 17:45.57 | *** join/#asterisk voxter (n=voxter@76.77.95.2) |
| 17:46.17 | [TK]D-Fender | dr_gogeta86: http://asterisk.name/asterisk/0596009623/asterisk-chp-4-sect-4.html |
| 17:46.25 | [TK]D-Fender | dr_gogeta86: http://www.google.ca/search?hl=en&q=Configure+FXO+port+with+zapata&btnG=Google+Search&meta=&aq=f&oq= |
| 17:46.39 | *** join/#asterisk omer (n=_omer@119.152.49.147) |
| 17:47.35 | *** join/#asterisk machoman48 (n=machoman@89.203.164.69) |
| 17:47.57 | omer | Do I need to install the Asterisk-Addons version similiar to my Asterisk verion??? or I can use any Asterisk Addons verion? it doesn;t matter which version of asterisk i have..???? |
| 17:48.48 | taylorS | SuprSluG: thanks for your help. i can get in with factory reboot, using the default password and reconfigured the phone. i've narrowed it down to a problem connecting to the boot server (probably uname/passwd since the IP is correct). Have to wait for the guy with pword... oh well. thanks agin for your help! |
| 17:49.17 | SuPrSluG | np |
| 17:51.51 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 17:56.35 | [TK]D-Fender | omer: of course it matters. |
| 17:56.56 | [TK]D-Fender | omer: Look at the relative release date of the major versions of the core vs addons for compatibility |
| 17:59.06 | dr_gogeta86 | [TK]D-Fender, i didn't see the incoming call |
| 17:59.36 | [TK]D-Fender | dr_gogeta86: and I don't see your configs. PASTEBIN is your friend. |
| 17:59.38 | [TK]D-Fender | ~pb |
| 17:59.38 | 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/ |
| 17:59.40 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 18:00.45 | dr_gogeta86 | [TK]D-Fender, my zaptel.com |
| 18:00.49 | dr_gogeta86 | http://pastebin.ca/1405127 |
| 18:02.13 | dr_gogeta86 | zapata.conf |
| 18:02.16 | dr_gogeta86 | http://pastebin.ca/1405128 |
| 18:02.36 | dr_gogeta86 | [TK]D-Fender, |
| 18:02.43 | dr_gogeta86 | here to you |
| 18:03.04 | [TK]D-Fender | dr_gogeta86: ... dialplan <- |
| 18:03.44 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-182-243-rbry-bb-1.dynamic.isadsl.co.za) |
| 18:04.13 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:04.28 | dr_gogeta86 | exten => _2.,1,Dial(Zap/g4/${EXTEN:1},20,tr) |
| 18:04.28 | dr_gogeta86 | exten => _2.,n,Hangup |
| 18:04.41 | dr_gogeta86 | but i didnt see any incoming call in the log |
| 18:05.37 | [TK]D-Fender | dr_gogeta86: and I don't see an appropriate exten for incoming calls to land on. |
| 18:05.37 | omer | [TK]D-Fender : how do I know which ADDONS is for my Asterisk Version??? I will really appreciate if you could explain a little bit....I really dont want my asterisk box to get f'd because of version shits... |
| 18:05.42 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-1ebc6c3bcf6d4c4e) |
| 18:05.51 | [TK]D-Fender | omer: What are you running now? |
| 18:06.00 | omer | asterisk-1.2.32 |
| 18:06.20 | dr_gogeta86 | [TK]D-Fender, sorry |
| 18:06.22 | dr_gogeta86 | exten => s,1,Answer() |
| 18:06.22 | dr_gogeta86 | exten => s,n,Echo(exten => s,1,Answer() |
| 18:06.22 | dr_gogeta86 | exten => s,n,Echo |
| 18:06.40 | [TK]D-Fender | dr_gogeta86: PASteBIN |
| 18:07.02 | [TK]D-Fender | omer: latest 1.2 addons |
| 18:07.25 | omer | thanks !! |
| 18:07.31 | omer | you have saved my box :) |
| 18:07.55 | omer | so I can install any 1.2.X addons ?? right |
| 18:07.59 | omer | ohh latest one.. |
| 18:08.01 | omer | I got it.. |
| 18:08.11 | omer | thanks....bye |
| 18:09.32 | dr_gogeta86 | [TK]D-Fender, and then |
| 18:18.08 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 18:19.53 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-4f45f05d87507526) |
| 18:20.51 | *** join/#asterisk GameGamer43 (n=GameGame@cpe-66-67-181-247.rochester.res.rr.com) |
| 18:27.09 | *** join/#asterisk awkfu (n=awkfu@66.162.90.56) |
| 18:33.31 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 18:39.06 | *** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk) |
| 18:40.10 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 18:41.05 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 18:41.32 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 18:41.58 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 18:42.19 | *** join/#asterisk lasko_ (n=lasko@70.102.15.210) |
| 18:43.39 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:48.41 | *** join/#asterisk dni (n=dniz0r@adsl-074-169-015-252.sip.mia.bellsouth.net) |
| 18:50.18 | dni | hello all,. could someone take a quick glance at these 8 lines from my inbound_menu context,. for somereason after it plays all the menu options,. it disconnects the call immediately. http://pastebin.com/m34db18f9 |
| 18:51.19 | dni | should i remove tha Hangup() from each 'step' ? |
| 18:52.15 | *** join/#asterisk rupa (i=rupa@99.180.116.104) |
| 18:52.24 | *** part/#asterisk juanIMP (n=Juancho@190.26.210.241) |
| 18:52.52 | [TK]D-Fender | dni: No doubt its because you did not set : autofallthrough=no under [globals] and you are not using WatExten. |
| 18:53.06 | [TK]D-Fender | sniSomething tells me you've been reading too many 1.0 guides |
| 18:53.20 | dni | [TK]D-Fender, just googline trying to figure everything out :) |
| 18:53.28 | dni | thanks for the feedback im going to revise it now |
| 18:55.12 | dni | [TK]D-Fender, i just read after 1.4 autofallthrough defaults to yes |
| 18:55.45 | [TK]D-Fender | dni: Hence the reason it cuts you off instantly. "s" runs out an dit has no reason to wait for input like it used to in to 1.0 days |
| 18:56.06 | dni | got ya |
| 18:56.06 | dni | thanks |
| 18:56.37 | *** part/#asterisk rupa (i=rupa@99.180.116.104) |
| 19:02.56 | beek | Do Polycom phones have a "screen-saver" type operation? When I create a background image of the appropriate size the icons overlay it. I was wondering if there is a mode that leaves just the background. |
| 19:03.51 | *** part/#asterisk machoman48 (n=machoman@89.203.164.69) |
| 19:06.35 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 19:08.47 | *** join/#asterisk neurosys (n=vinix@173.9.134.209) |
| 19:10.38 | [TK]D-Fender | beek: nope |
| 19:11.13 | beek | [TK]D-Fender: I was afraid someone would say that. I've been reading all of the Polycom docs and wondered why they insist on calling it the "idle image." |
| 19:11.23 | beek | Thanks |
| 19:13.22 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 19:13.24 | [TK]D-Fender | beek: because that what display on idle. Your indicators are still necessary on idle to indicate things loikke MWI, forwarding, missed calls, etc |
| 19:14.31 | *** join/#asterisk LakeSolon (n=blake@96-42-127-243.dhcp.roch.mn.charter.com) |
| 19:15.18 | beek | [TK]D-Fender: That makes sense, of course. It's just a bit disappointing that most of the screen real estate is being eaten by the button tags. |
| 19:15.58 | beek | [TK]D-Fender: I have a growing collection of phones here: the IP501, IP450 and IP330. |
| 19:16.13 | [TK]D-Fender | beek: Shame on them for ensuring the visibility of important status indication! |
| 19:16.40 | beek | [TK]D-Fender: Absolutely. What the hell does Polycom think it's making? |
| 19:16.43 | *** join/#asterisk jnials (n=jnials@cuervo.unwiredbuyer.com) |
| 19:18.20 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 19:32.53 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 19:45.58 | *** join/#asterisk kuku1 (n=ingo@c-98-227-117-244.hsd1.il.comcast.net) |
| 19:47.16 | kuku1 | We have upgraded from 1.2 to 1.4 and are experiencing some weird sound issues with asterisk. We have removed all the modules from the modules directory prior to compiling 1.4. Any suggestions as to what the issue might be ? I'm happy to provide any information required to diagnose this. We had this running for 1.5 years fine with version 1.2 |
| 19:47.49 | [TK]D-Fender | kuku1: Perhaps you could elaborate on "issues" |
| 20:02.21 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 20:05.18 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:08.55 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 20:09.57 | *** join/#asterisk plq (n=plq@88.250.169.4) |
| 20:11.31 | *** join/#asterisk telecos (n=sergio@213.167.219.87.dynamic.jazztel.es) |
| 20:14.38 | timeshell_atwork | what's the best wifi sip phone to use with asterisk? |
| 20:20.10 | *** join/#asterisk Victor_Yure_ (n=victor@unaffiliated/victoryure/x-837844) |
| 20:20.14 | [TK]D-Fender | timeshell_atwork: All pretty much suck |
| 20:21.37 | *** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7) |
| 20:24.04 | [TK]D-Fender | checkout time, BBIAB |
| 20:25.47 | Katty | hi. |
| 20:26.47 | carrar | Firefox IRC, thats serious stuff!! |
| 20:27.05 | carrar | Hi none-hugable Katty! |
| 20:27.09 | Katty | hello what is asterisk |
| 20:27.20 | carrar | It's a web browser |
| 20:27.28 | Katty | what is web browser please? |
| 20:27.41 | carrar | It's a tube viewer |
| 20:27.47 | Katty | you fail. |
| 20:27.48 | Katty | eppigy: DAVE |
| 20:28.00 | *** join/#asterisk wpbrown (n=wpbrown@wh-gtw-0001.woolfharris.com) |
| 20:28.52 | Katty | Qwell: Gwendolen the Noble |
| 20:29.20 | Katty | Qwell: next week, Matron Gwynne |
| 20:29.21 | Qwell | tsk tsk |
| 20:29.35 | Katty | tsker. |
| 20:29.40 | Katty | you know you wanna be Patron Qwell |
| 20:30.18 | Katty | hugs JayTee52 |
| 20:30.43 | Katty | today's been incredibly uneventful. |
| 20:30.52 | Katty | everything appears to be working. |
| 20:30.57 | Katty | :< |
| 20:31.42 | Qwell | Katty: you need to break something then |
| 20:32.07 | Katty | eggs benedict for dinner! |
| 20:35.26 | JayTee52 | oooh, that's a rich dinner |
| 20:35.35 | JayTee52 | hugs Katty |
| 20:35.52 | beek | Hello JayTee52 |
| 20:36.06 | JayTee52 | I'm in here twice cuz I forgot to logoff at home :-) |
| 20:36.09 | JayTee52 | hi beek |
| 20:36.23 | *** join/#asterisk goofy03 (n=kvirc@31.102.203-77.rev.gaoland.net) |
| 20:36.29 | goofy03 | Hi |
| 20:36.51 | JayTee52 | typical first day back from vacation, a total zoo at a total zoo. sounds kinda redundant |
| 20:37.30 | Katty | oh? vacation? |
| 20:37.31 | Katty | where did you go? |
| 20:37.41 | JayTee52 | I went to Indianapolis. |
| 20:37.48 | Katty | ooooh! |
| 20:37.52 | Katty | sounds exciting! |
| 20:37.55 | Katty | did you post pictures yet? |
| 20:37.56 | JayTee52 | from Indianapolis |
| 20:38.19 | goofy03 | I get "Asterisk died with code 1." in an infinite loop and nothing wrong in full log |
| 20:38.24 | JayTee52 | the 5 and a half mile drive was gruelling |
| 20:38.39 | carrar | Anyone order theirs yet? http://www.approvedgasmasks.com/suit-TK640.htm |
| 20:38.42 | beek | Do I detect sarcasm? |
| 20:38.46 | goofy03 | dont know what to do |
| 20:39.00 | JayTee52 | maybe just a teensy bit |
| 20:39.09 | beek | goofy03: Did this work before? |
| 20:39.58 | JayTee52 | basically it was a stay at home vacation, where I just chilled and didn't have to deal with TSA idiots, baggage claims, rude waiters, overpriced hotels, cranky rental cars and shitty weather someplace else. |
| 20:40.13 | goofy03 | yep with debian init script |
| 20:40.37 | beek | goofy03: Start it from a command line and see if you can see what it's malfunction is. |
| 20:41.16 | goofy03 | heu sorry i try to launch with freepbx |
| 20:41.45 | *** join/#asterisk Anth8708 (n=SaiSoma@client105.jdcc.edu) |
| 20:42.33 | beek | JayTee52: You heading to Astricon this year? |
| 20:42.55 | JayTee52 | probably not, I'm more of an old hack than a dev |
| 20:43.22 | beek | I'm kicking it around. |
| 20:43.44 | JayTee52 | I'd have to pay for it myself and I'd never make that money back so it's not likely |
| 20:44.05 | beek | I'm going to pay for it myself, as well. |
| 20:44.06 | *** join/#asterisk Anth8708 (n=SaiSoma@client105.jdcc.edu) |
| 20:44.59 | JayTee52 | some people just can't make up their mind who they are |
| 20:45.07 | Anth8708 | sorry, new chat client |
| 20:45.14 | beek | Or their schizophrenic |
| 20:45.20 | beek | s/their/they're/ |
| 20:45.48 | JayTee52 | you're never alone with a schizophenic |
| 20:46.30 | Anth8708 | quick question. any way to use a nortel analog line card to trunk? i just need to get the card to send the called number to asterisk so the tdm800p can pick it up |
| 20:46.54 | JayTee52 | Anth8708, yep, that'll work |
| 20:47.24 | JayTee52 | assuming you're plugging the line from the Nortel analog TN to an FXO port. |
| 20:47.51 | Anth8708 | yup. i am. right now, asterisk sees the CID from the nortel, but doesn't see the called number |
| 20:47.53 | JayTee52 | but you won't get CID |
| 20:48.00 | Anth8708 | so i can't route |
| 20:48.11 | JayTee52 | you have a Class modem card for CID? |
| 20:48.13 | Anth8708 | JayTee52: so it's an either or? |
| 20:48.17 | Anth8708 | JayTee52 yes |
| 20:49.08 | Anth8708 | JayTee52: just don't know how to tell the nortel to send the called number to asterisk:(. is it in the cls perhaps? |
| 20:49.30 | JayTee52 | Anth8708, no you have to either assign a 1 to 1 analog to FXO line or use an IVR since the nortel is dialing an extension on it's own switch and is just going offhook on the TN |
| 20:50.33 | Anth8708 | JayTee52: rgr. so can i reconfig the normal analog line as a "trunk?" |
| 20:50.54 | JayTee52 | Anth8708, it will never behave like a true trunk since it isn't and it is not programmed as a trunk in BARS/NARS so you don't have a steering code to make it go offhook and pass any additional digits over it. |
| 20:51.39 | Anth8708 | JayTee52: so i'm likely better off creating bogus dialing code and routing with the nortel over a pri card perhaps then? |
| 20:52.04 | eppigy | hi Katty ^_________^ |
| 20:52.20 | JayTee52 | so if you have a one to one correspondence of analog numbers on Nortel you want to route to similar number on Asterisk you can just have it setup to answer |
| 20:52.46 | JayTee52 | Anth8708, yeah, you'd get dialed digits and CID over PRI but only from a DTI/PRI not a Line Side T1 |
| 20:53.29 | JayTee52 | Anth8708, do you currently have PRI's to your telco? |
| 20:53.31 | Anth8708 | JayTee52: I have 1-to-1 setup now, but wanted to use the voicemail on asterisk. guess i'd be better off (for now anyway) still using the existing VM in the nortel |
| 20:53.42 | JayTee52 | you would |
| 20:54.02 | Anth8708 | JayTee52: we have an incoming PRI to our option 11, yup. and an extra PRI card that was configured for routing to another cabinet on the other campus |
| 20:54.02 | beek | Anth8708: Does the Nortel sit between the PSTN and Asterisk? |
| 20:54.22 | Anth8708 | beek: yes it does. for now. this fall, we sell the nortel and move completely to asterisk |
| 20:54.32 | beek | Anth8708: Cool. |
| 20:54.34 | Anth8708 | beek: but for now, we use hybrid |
| 20:54.37 | JayTee52 | I migrated all our phones from a Nortel 11C to Asterisk by putting Asterisk in between the telco and the Nortel on the OUTBOUND PRI span |
| 20:55.25 | beek | Anth8708: I did what JayTee52 did for my Iwatsu. |
| 20:55.34 | JayTee52 | and using remote call forwarding with a station control password to forward phantom analog TNs with nortel DNs assigned to route to a 9555-XXXX number. |
| 20:56.16 | Anth8708 | JayTee52: hmm . interesting. i might could just let asterisk handle the pri completely to the pstn and route any unknown calls directly to the nortel and setup the pri between the two systems. let the nortl use it's existing pri (was to telco) to talk to the asterisk box |
| 20:56.33 | Anth8708 | then, technically, i could migrate at leisure |
| 20:56.37 | JayTee52 | which hit's the incoming context for that PRI span on Asterisk and hits a set of pattern matches, most for outbound but one for the 555 to strip the 555 and route the call with the last 4 digits to an internal context |
| 20:56.56 | JayTee52 | Anth8708, yep I migrated over a period of 8 months |
| 20:57.07 | Anth8708 | Thanks guys |
| 20:57.16 | Anth8708 | i may have to try that this weekend as we complete our testing |
| 20:57.18 | Anth8708 | awesome |
| 20:57.33 | JayTee52 | Anth8708, I used phantom TN's for the analog and also to reprogram for the digital sets I rerouted |
| 20:57.56 | Anth8708 | JayTee52: *nod* |
| 20:58.26 | JayTee52 | two things, the T1 cable from the Nortel to the * is a crossover. the T1 cable from Asterisk to the CSU is a straight through. |
| 20:58.40 | Anth8708 | JayTee52: gotcha. |
| 20:59.07 | JayTee52 | and the Nortel in almost every case and every switch I've worked on is set as CPE so the span on * that connects is set to pri_net and provides timing. |
| 20:59.32 | Anth8708 | JayTee52: *nod* makes sense. we're emulating the pstn at that point |
| 20:59.54 | JayTee52 | beek and I hang here often so if you're trying it out drop back here and look for me. I'm usually here without the 52 at the end of my nick |
| 21:00.17 | Anth8708 | JayTee52: rgr that. i added this channel to my normal haunts now. thanks |
| 21:00.27 | JayTee52 | no problem |
| 21:00.40 | Anth8708 | brb. going to look at the physical routing of cabling |
| 21:02.21 | *** join/#asterisk Gnutoo (n=gnutoo@host126-144-dynamic.54-79-r.retail.telecomitalia.it) |
| 21:03.00 | Gnutoo | hello, I don't know if it's the best channel for asking but I'm looking for a sip client or library |
| 21:03.12 | Gnutoo | it should have video |
| 21:03.14 | Qwell | Gnutoo: Asterisk is a "sip client" |
| 21:03.20 | Gnutoo | wow |
| 21:03.25 | Gnutoo | great |
| 21:03.32 | Qwell | asterisk is lots of things |
| 21:03.40 | Gnutoo | I use it as server actually |
| 21:03.53 | JayTee52 | I use mine to run my Winnebago |
| 21:04.06 | Gnutoo | can it display your video on a display and the contact's video on another display? |
| 21:05.10 | Gnutoo | i'll check wich version are avaliable in openembedded |
| 21:05.48 | JayTee52 | quittin time, be back later |
| 21:06.27 | *** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221) |
| 21:11.55 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:14.31 | Katty | eppigy: ohhaideriwasafk |
| 21:18.55 | eppigy | Katty: HERRO |
| 21:18.59 | eppigy | what is for dinner |
| 21:19.01 | eppigy | I am starving |
| 21:19.27 | Katty | idk |
| 21:19.30 | Gnutoo | how can I use asterisk as a "sip client" ? |
| 21:19.31 | Katty | my bff, jill? |
| 21:19.50 | Katty | eppigy: burritos maybe. |
| 21:20.14 | Katty | eppigy: possible leftover pork chop apple pie and stuffing. |
| 21:21.29 | [TK]D-Fender | Gnutoo: meaning? |
| 21:21.32 | Katty | eppigy: there's also some pasta in the fridge. |
| 21:21.42 | Katty | eppigy: which could be mixed with a can of tuna ands ome cheese. |
| 21:21.54 | war9407 | I have multiple GoToIfs() |
| 21:22.01 | war9407 | How come only the first one is evaluated? |
| 21:22.12 | war9407 | ah |
| 21:22.16 | war9407 | I need to say Function:n |
| 21:22.23 | war9407 | then it should go to the next line |
| 21:22.27 | war9407 | and put it on the LASt match |
| 21:22.31 | war9407 | where I want it to go it none of them match |
| 21:22.39 | Gnutoo | [TK]D-Fender, I was told by Qwell that asterisk could be used as a sip client...I've a device with 2 320x240 screens and I'd like to try to make a sip client that uses both screens: one for each video(one for mine and one for my contact) |
| 21:23.09 | Gnutoo | else I'm looking for a good client or lib |
| 21:23.34 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 21:23.58 | eppigy | Katty: YES |
| 21:24.03 | eppigy | oh man |
| 21:24.06 | eppigy | I am so hungry |
| 21:24.38 | Katty | maybe you should have a snack. |
| 21:25.10 | [TK]D-Fender | Gnutoo: * is not a soft-phone |
| 21:25.24 | [TK]D-Fender | Gnutoo: and what is this other device you're talking about? |
| 21:25.42 | Qwell | sounds like a tuxphone or something |
| 21:25.48 | Gnutoo | ok thanks...it's the bug device from buglabs: |
| 21:26.01 | Gnutoo | http://www.buglabs.net/ |
| 21:26.27 | Gnutoo | I've also an openmoko but it doesn't have 2 screens |
| 21:26.47 | eppigy | there are no snacks here :[ |
| 21:26.57 | eppigy | I am about to dip out of work though |
| 21:27.26 | [TK]D-Fender | Gnutoo: * is not a video-conference tool |
| 21:27.34 | Gnutoo | ok thanks |
| 21:27.39 | Gnutoo | lol |
| 21:28.10 | Gnutoo | so what could I use? |
| 21:28.44 | goofy03 | if i dont load module dynamically asterisk work how can i find the faulty mod ? plz |
| 21:28.48 | [TK]D-Fender | Gnutoo: Something else |
| 21:28.50 | *** join/#asterisk deeperror (n=deeperro@adsl-99-33-114-255.dsl.sfldmi.sbcglobal.net) |
| 21:29.22 | Gnutoo | ok thanks |
| 21:29.31 | Gnutoo | I'll ask in another channel |
| 21:29.48 | deeperror | Any clues why when performing 3-way calls the channel instance would always show 1-1 and not show 1-2 when putting the caller on hold? |
| 21:30.52 | *** join/#asterisk telecos (n=sergio@213.167.219.87.dynamic.jazztel.es) |
| 21:33.34 | Katty | my hunger is eating me alive |
| 21:33.36 | Katty | from the inside |
| 21:33.46 | wpbrown | Tacos |
| 21:33.50 | [TK]D-Fender | deeperror: pastebin...... |
| 21:34.15 | watchy | i fixed my issue the other day tk, the first problem was i'm retarded, the 2nd was i was using the wrong group |
| 21:40.49 | *** part/#asterisk Gnutoo (n=gnutoo@host126-144-dynamic.54-79-r.retail.telecomitalia.it) |
| 21:42.42 | war9407 | asterisk not passing caller id -> to phone with 1.4, but it did with 1.6 |
| 21:42.47 | war9407 | what to look for? |
| 21:42.54 | war9407 | watchy: btw, the watchyi used to know? @iniquity? |
| 21:43.07 | deeperror | www.pastebin.ca/1405401 |
| 21:43.38 | goofy03 | pbx.c: Already have an application 'Directory' |
| 21:44.11 | goofy03 | what is the problem ? |
| 21:44.19 | deeperror | [TK]D-Fender pastebin above |
| 21:46.45 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 21:46.53 | [TK]D-Fender | deeperror: I'd have liked to have seen the complete call... |
| 21:47.04 | deeperror | i'll get one tomorrow when the pbx is slower |
| 21:47.24 | deeperror | hard to get the full call on the one that isn't showing the instance properly |
| 21:47.46 | deeperror | but no setting that would change that that you know of off hand? |
| 21:47.56 | [TK]D-Fender | goofy03: You have multiple voicemail or Directory apps apps being loaded, which is usually because of DB varients. |
| 21:48.04 | [TK]D-Fender | goofy03: noload the ones you don't really need |
| 21:48.15 | [TK]D-Fender | deeperror: Nothing I know of |
| 21:48.19 | deeperror | ok thanks |
| 21:48.22 | deeperror | i'll try to get more info tomorrow |
| 21:48.25 | war9407 | [TK]D-Fender: any clues? |
| 21:48.26 | deeperror | thanks for your time |
| 21:48.27 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 21:48.52 | Katty | eppigy: ohohoh |
| 21:48.56 | Katty | eppigy: lookit the TIME |
| 21:49.07 | *** join/#asterisk ddickenson (n=ddickens@67-198-0-5.static.grandenetworks.net) |
| 21:49.15 | [TK]D-Fender | war9407: Like usual I've been shown nothing of value |
| 21:50.04 | war9407 | [TK]D-Fender: http://pastebin.com/m43373844 |
| 21:50.26 | [TK]D-Fender | war9407: keep going... |
| 21:50.27 | war9407 | [TK]D-Fender: I get the email from the script but when it rings the phone (ringphone), I do not see the caller id info on the phone (it worked with 1.6. branch) |
| 21:51.14 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
| 21:51.38 | war9407 | it just says "incoming call" on the phone (normally it showed the name & number/etc) |
| 21:51.50 | goofy03 | [TK]D-Fender: Where do u set up Directory apps ? |
| 21:51.57 | [TK]D-Fender | war9407: I don't see a CALL anywhere |
| 21:52.05 | war9407 | [TK]D-Fender: sec |
| 21:53.01 | [TK]D-Fender | goofy03: disable the ones you don't need in modules.conf |
| 21:53.23 | war9407 | [TK]D-Fender: http://pastebin.com/m11a1baae |
| 21:53.29 | *** join/#asterisk OuterSpace (n=qwerty20@190.223.163.13) |
| 21:53.37 | war9407 | # |
| 21:53.37 | war9407 | [Apr 27 17:52:40] DEBUG[24361] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP) |
| 21:53.37 | war9407 | hmm |
| 21:54.25 | goofy03 | there are too many and i dont know what i need |
| 21:54.35 | war9407 | [TK]D-Fender: is it something simple or ? |
| 21:54.48 | *** join/#asterisk ttl- (n=patrick@d5153AE82.access.telenet.be) |
| 21:54.51 | [TK]D-Fender | war9407: Where did all that crap come from? |
| 21:55.00 | [TK]D-Fender | war9407: this does not looks like * CLI |
| 21:55.07 | war9407 | [TK]D-Fender: the logs |
| 21:55.15 | [TK]D-Fender | warSCREW LOGS. CLI output only |
| 21:56.14 | war9407 | the output is the same |
| 21:56.49 | OuterSpace | hi, i installed asterisk last friday from svn and got the "stuck call on Voicemail" problem. I see bug is still open today. nothing going to happend to g729 licenses on upgrade rigth ? |
| 21:57.58 | OuterSpace | i used 4 g729 licenses, and i need to upgrade to solve the bug, the question is, should i have to register licenses again ? |
| 21:58.00 | [TK]D-Fender | war9407: war9407 No, it isn't |
| 21:58.11 | war9407 | [TK]D-Fender: when you have the verbose & debug set to 999 it is |
| 21:58.20 | war9407 | same exact msgs. |
| 21:58.26 | [TK]D-Fender | war9407: And you think SIP DEBUG is not important? |
| 21:58.54 | [TK]D-Fender | war9407: look at what you're really sending |
| 22:03.17 | war9407 | hum |
| 22:04.10 | kuku1 | [TK]D-Fender: regarding initial issue: <kuku1> We have upgraded from 1.2 to 1.4 and are experiencing some weird sound issues with asterisk. We have removed all the modules from the modules directory prior to compiling 1.4. Any suggestions as to what the issue might be ? I'm happy to provide any information required to diagnose this. We had this running for 1.5 years fine with version 1.2 |
| 22:05.41 | kuku1 | [TK]D-Fender: When calling, during the conversation, randomly, the call will have disruptions. sometimes I will talk, and all of a sudden, I will hear what I just said ( in a distorted way ), and then I feel like the other side just hears the repetition. |
| 22:05.42 | [TK]D-Fender | has clearly entered Planet Crack |
| 22:05.59 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 22:06.10 | kuku1 | [TK]D-Fender: The other side also states that the call breaks up. |
| 22:06.36 | kuku1 | this is only sip to sip, sip to iax I could not reporduce |
| 22:06.55 | war9407 | [TK]D-Fender: any other hints? basically the same configuration works under 1.6 |
| 22:07.07 | war9407 | [TK]D-Fender: the caller ID is passed to the regular telephone (analog) |
| 22:07.12 | war9407 | but with 1.4, nada |
| 22:07.59 | [TK]D-Fender | war9407: I'm not seeing the call or configs, and saying "it worked in 1.2" is a complete waste of time. Go provide something useful like i've had to ask you for half a dozen times already. |
| 22:08.27 | [TK]D-Fender | war9407: Caller-id does not work any less on 1.4 and something is screwed up and you're not looking where I'm telling you to. |
| 22:09.19 | kuku1 | [TK]D-Fender: I was getting notices regarding moho, so I disabled moho, but same issue. |
| 22:09.28 | [TK]D-Fender | moho? |
| 22:09.29 | war9407 | [TK]D-Fender: you want to see sip.conf? |
| 22:09.39 | OuterSpace | if i upgrade asterisk i keep g729 licences ? |
| 22:09.48 | [TK]D-Fender | war9407: I want to see the actual call the way I asked for it. REPEATEDLY |
| 22:10.07 | war9407 | [TK]D-Fender: I showed it to you, what debug/verbosity setting you want to see? |
| 22:10.14 | war9407 | at 999 its the same as the asterisk.log file |
| 22:10.51 | [TK]D-Fender | war9407: i want to see the god-damned sip packet in in it &^#$ing entirety with SIP DEBUG enabled at CLI. What don't you get? |
| 22:10.52 | *** join/#asterisk rue_mohr (n=rue@24.207.122.10) |
| 22:10.59 | [TK]D-Fender | wareVerbose does not mean SHIT to this |
| 22:11.10 | rue_mohr | how do I turn off dtmf muting? |
| 22:11.20 | [TK]D-Fender | war9407: Now go to * CLI, enable SIP DEBUG and pastebin the entire damn call |
| 22:11.28 | war9407 | k |
| 22:13.17 | *** part/#asterisk jnials (n=jnials@cuervo.unwiredbuyer.com) |
| 22:14.06 | war9407 | [TK]D-Fender: http://pastebin.com/m199deb65 |
| 22:15.08 | rue_mohr | you know how asterisk so politely mutes dtmf to the remote party? |
| 22:16.10 | rue_mohr | our people want to be able to use the menu options on other peoples phone systems, so I need asterisk to not mute dtmf |
| 22:16.32 | Qwell | rue_mohr: no, that isn't what you need. |
| 22:16.33 | [TK]D-Fender | war9407: sip config now |
| 22:16.50 | rue_mohr | oh, k, do I need everyone to stop using those pesky menus? |
| 22:17.12 | Qwell | you need to setup your SIP DTMF config properly |
| 22:17.14 | rue_mohr | I'll start phoning them now I'll start at 001-001-0001 |
| 22:17.49 | rue_mohr | ok how do I set it so it dosn't mute the dtmf on the remote side |
| 22:18.02 | war9407 | sec |
| 22:18.03 | rue_mohr | I know asterisk does this, cause it happens on my all-analog system at home |
| 22:18.38 | Qwell | Fix your dtmfmode |
| 22:18.50 | war9407 | [TK]D-Fender: http://pastebin.com/m682c6c45 |
| 22:18.55 | Qwell | and don't get sarcastic with me. |
| 22:19.48 | rue_mohr | Qwell, it works on some systems, cause there hmm, wait a sec... |
| 22:20.18 | rue_mohr | hmm I just tried calling the shop on my cell and having them push a digit, the dtmf does come thru |
| 22:20.24 | rue_mohr | my bad, |
| 22:20.40 | rue_mohr | so why cant a bunch of systems properly detect our dtmf |
| 22:20.44 | Qwell | Fix your dtmfmode |
| 22:21.32 | rue_mohr | what setting do you suggest I use? |
| 22:21.41 | Qwell | The one that matches your provider. |
| 22:21.57 | rue_mohr | my provider is a digium card |
| 22:22.01 | [TK]D-Fender | war9407: From: "Cell Phone VA" <sip:7033422193@192.168.168.254>;tag=as44cced73 |
| 22:22.03 | [TK]D-Fender | To: <sip:line1@192.168.168.246:5060> |
| 22:22.06 | rue_mohr | in the machine that runs the phonesystem |
| 22:22.09 | Qwell | then the one that matches your phones. |
| 22:22.21 | [TK]D-Fender | war9407: outbound packet has it. Go look at your device. * has done its job |
| 22:22.39 | war9407 | [TK]D-Fender: ah |
| 22:22.44 | rue_mohr | our menus work fine, but when we call other peoples phone systems, they aren't detecting our dtmf |
| 22:23.01 | rue_mohr | its a far end problem |
| 22:23.29 | rue_mohr | I think its related to gain and distortion, cause when she pushed the button the audio dtmfcame through pretty rough |
| 22:24.20 | war9407 | [TK]D-Fender: ok, will look, thx. |
| 22:24.20 | Qwell | are your gains still at 94? |
| 22:24.20 | [TK]D-Fender | rue_mohr: With your perfectly sane settings? Say it ain't so! |
| 22:24.20 | Qwell | walks away |
| 22:24.20 | rue_mohr | no, our gains are set to "sounds good" |
| 22:24.20 | rue_mohr | as you suggested |
| 22:24.24 | Qwell | and those values...are...what? |
| 22:24.28 | rue_mohr | which is somewhere between -3 and 16 db |
| 22:24.29 | seanbright | Qwell: e-cig update |
| 22:24.34 | [TK]D-Fender | rue_mohr: You've been watching too much Spinal Tap |
| 22:24.39 | Qwell | seanbright: coughing up crap today |
| 22:24.44 | Qwell | tar ftl |
| 22:24.50 | rue_mohr | cause the agc ont eh phones makes any adjustments pointless |
| 22:24.59 | seanbright | Qwell: cleansing ftw, however. |
| 22:25.05 | seanbright | the "toxins" |
| 22:25.08 | Qwell | seanbright: yeah.. |
| 22:25.15 | seanbright | i hate people who talk about "toxins" |
| 22:25.17 | rue_mohr | keep in mind I adjusted things because I HAD TO |
| 22:25.18 | seanbright | hates himself |
| 22:25.24 | seanbright | runs away |
| 22:25.30 | rue_mohr | because it did not work with the defaults |
| 22:25.31 | Qwell | rue_mohr: start over with the gains. put them at 0, and make them work. |
| 22:25.44 | Qwell | SMALL adjustments. |
| 22:25.53 | Qwell | of course a gain of 16 is going to break things |
| 22:26.00 | rue_mohr | I did, the agc rendered small changes pointless |
| 22:26.08 | Qwell | so don't use agc |
| 22:26.30 | rue_mohr | I dont think the sip phones have an 'off' for the agc |
| 22:26.32 | *** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com) |
| 22:26.36 | Qwell | "it" doesn't work with the defaults? |
| 22:26.39 | Qwell | what is "it"? |
| 22:27.19 | rue_mohr | everyone in the office says "I cant hear the person on the other side of the line, even when its tured all the way up I still have to ask them to keep repeating themselfs" |
| 22:28.42 | rue_mohr | anyone heard of a 'cleanser' script that can turf all the stuff out of the polycom sip.cfg that is for phones I dont have? |
| 22:30.38 | rue_mohr | can anyone confirm if its true the 1mw is not 1mw? |
| 22:31.40 | Qwell | 1mw what? |
| 22:32.14 | rue_mohr | there is a tone function to generate a 1mw signal, iirc there was a newsgroup posting saying not to use it cause its wrong somehow |
| 22:32.31 | rue_mohr | I would much like dahdi_monitor to have a db scale on it |
| 22:32.43 | rue_mohr | I'v determines the bar graph means nothing |
| 22:32.46 | rue_mohr | d |
| 22:33.29 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 22:34.30 | rue_mohr | it would also be great to have a realtime level monitor for sip calls |
| 22:35.19 | rue_mohr | does dahdi_monitor work directly with the card? |
| 22:36.50 | deeperror | Anyone ever seen this? "Building conference on call on Zap/38-1 and Zap/38-1" when it should be 38-2 on one of the channels? |
| 22:40.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 22:44.28 | rue_mohr | http://www.mail-archive.com/asterisk-users@lists.digium.com/msg214182.html oh look, I'm not the only one in the world with this problem |
| 22:52.18 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 23:01.35 | rue_mohr | is this channel logged anywhere people searching cn find information from it? |
| 23:02.59 | rue_mohr | the difference seems to be: polycom changed the two following values in their config files: voice.gain.tx.analog.handset from 6 to 12 and voice.gain.tx.analog.preamp.handset from 23 to 14 |
| 23:03.27 | rue_mohr | I had dialed my chassis volume up, which I thought was an overall gain, but I belive now to just be the keypad |
| 23:03.31 | rue_mohr | (dtmf) |
| 23:06.54 | *** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose) |
| 23:09.34 | rue_mohr | hmm what is the number for my local 1mw |
| 23:12.00 | rue_mohr | anyone know a keyword I can use to find that directory of numbers with the middle inger logo? |
| 23:12.29 | crunge | in a dialplan, is it possible to loop over a list of values in a variable? |
| 23:12.53 | rue_mohr | http://www.hackcanada.com/telco/telus_numbers.html found it |
| 23:16.03 | crunge | loop with cut =D |
| 23:19.53 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
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| 23:23.27 | *** mode/#asterisk [+o russellb_] by ChanServ |
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| 23:24.22 | rue_mohr | I still need to know what is doing the idle muting for the phones, everyone hates it |
| 23:25.44 | rue_mohr | its asterisk, its not the sip sets because its happening with an analog set I just tried |
| 23:26.02 | rue_mohr | so, how many places in asterisk will do idle mute |
| 23:26.38 | russellb_ | Just meetme, and even there not by default |
| 23:27.01 | rue_mohr | odd, this is just a dahdi-> dahdi call |
| 23:27.19 | phix | dahdi == zap? |
| 23:27.23 | rue_mohr | yes |
| 23:27.39 | rue_mohr | and something in there is doing idle mute |
| 23:27.45 | rue_mohr | wonder if its the hwec |
| 23:28.20 | rue_mohr | echocancelwhenbridged=yes <-- should that be no? |
| 23:28.22 | phix | could be |
| 23:28.38 | phix | rue_mohr: depends if you want to send faxes |
| 23:28.47 | rue_mohr | I do, and I cant |
| 23:28.52 | rue_mohr | to some people |
| 23:28.56 | phix | rue_mohr: so it should be no then |
| 23:29.00 | rue_mohr | ah! |
| 23:29.02 | rue_mohr | cool |
| 23:29.06 | rue_mohr | nice to get that off my list |
| 23:29.55 | phix | if you have a fax machine hooked up to your TDM card and you send or receive a fax via TDM as well then you have two zap channels bridged |
| 23:30.14 | phix | you dont want echo cancelling on if it is a fax as it fucks shit up :) |
| 23:30.24 | rue_mohr | yea |
| 23:30.39 | rue_mohr | hmm are all the hwec settings in the chan_dahdi.conf file then? |
| 23:30.54 | rue_mohr | ah I can use this as a test |
| 23:30.55 | phix | ja |
| 23:33.15 | rue_mohr | bingo |
| 23:33.22 | rue_mohr | oh.... |
| 23:34.16 | rue_mohr | ok, its the hwec, where the *&^&^$% is the switch!? |
| 23:36.25 | rue_mohr | # set the transmit quiet dropoff burst time in milliseconds: |
| 23:36.26 | rue_mohr | #bursttime=234 |
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| 23:47.23 | voxter | Qwell: ping? |
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