| 00:00.16 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 00:01.03 | jhirley | hey folks on the smaller side, what is a good 4/8 port fxo gateway and why do like them ? |
| 00:07.41 | p3nguin | Okay, now how do we add cable TV to this package? |
| 00:07.59 | *** join/#asterisk Ickmund (n=magnus@ada-bcn-fw01.adamoeurope.com) |
| 00:08.25 | bmoraca_work | i am not familiar with that, though i do know that a number of platforms exist |
| 00:08.46 | *** join/#asterisk coppice (n=chatzill@185.204.17.210.dyn.pacific.net.hk) |
| 00:08.56 | carrar | jhirley, Audio codes |
| 00:09.01 | bmoraca_work | honestly, i wouldn't bother with it, because you're not going to be able to offer a lot of the digital cable features users are going to want (on demand, that sort of things) |
| 00:09.10 | p3nguin | I know some small rural ISPs around here offer a triple-play over a single copper line to the house. In the house, the wire goes to the cable box, and then out to your computer. |
| 00:09.23 | p3nguin | They offer all that stuff. |
| 00:09.46 | p3nguin | uh, single copper pair |
| 00:10.03 | bmoraca_work | that's how uVerse works...it's IPTV |
| 00:10.12 | bmoraca_work | you might be able to get in touch with a provider for that |
| 00:10.14 | p3nguin | Actually, I've only seen it in rural ISPs. |
| 00:10.27 | *** join/#asterisk TheDavidFactor-H (n=chatzill@nc-71-0-16-133.dhcp.embarqhsd.net) |
| 00:10.37 | bmoraca_work | that's how i get uVerse...single copper pair into an RG, and i get my TV, internet, and phone through it |
| 00:11.22 | jhirley | you may also want to look at fios vs uverse they are both being deployed in the us |
| 00:11.34 | p3nguin | He's referring to it as VDSL. Does that sound like the right term? |
| 00:11.44 | bmoraca_work | you would probably be better off just striking a resale bargain with the local cable company for this...from my experience, this is far more expensive. |
| 00:11.52 | jhirley | uverse is supposed to be 18 megs fios is 88megs depending on who you speak with. |
| 00:12.02 | bmoraca_work | p3nguin, yes, that's what uVerse is...fiber to the node and VDSL to the premesis |
| 00:12.24 | bmoraca_work | p3nguin, you can achieve the same thing over ADSL2+ for the most part, but those Adtran boxes support VDSL too |
| 00:12.45 | bmoraca_work | VDSL modems are far less common, though |
| 00:13.23 | p3nguin | The 1248 and 1200F support VDSL? |
| 00:13.29 | bmoraca_work | TV isn't something you want to mess around with...if you implement it wrong, you can be responsible for big money. i'd let the TV company handle the TV portion |
| 00:13.30 | bmoraca_work | yes |
| 00:13.47 | carrar | iptv been around forever! |
| 00:13.48 | carrar | http://pics.osburn.com/photo/5521/original |
| 00:13.59 | jhirley | you may also want to look at your back end software, vdsl my get you the bandwidth but you don't want to be like att and have all sorts of backend problem with the user interface. |
| 00:14.28 | bmoraca_work | jhirley, AT&T fixed all those issues about 9 months ago. i haven't had a problem at all since that update |
| 00:16.11 | bmoraca_work | p3nguin, sorry, the 1248 doesn't support VDSL...i believe that's the 1148 that does |
| 00:16.12 | jhirley | bmoraca_work: all "those" issues are on going and seem to be resolved on a case by case basis. Milage does vary from on person to another but the service as a whole is a great product. |
| 00:17.31 | jhirley | you guys may want to check out www.hatterasnetworks.com they had some nice boxes that could push 50 megs over multiple pairs. aka metro ethernet or optiwan |
| 00:18.05 | jhirley | but you need a clec license to use them since the pair are un-bundled network elements. |
| 00:22.05 | bmoraca_work | voice and data are easy to provide...cable internet is significantly more difficult...i'd leave that to the people who already have that infrastructure |
| 00:22.09 | bmoraca_work | er...cable tv |
| 00:23.00 | jhirley | er ... uverse , lol |
| 00:27.32 | *** join/#asterisk n3hxs (n=HAMming@pool-96-241-151-251.washdc.fios.verizon.net) |
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| 00:33.10 | *** join/#asterisk lupine_85 (n=quassel@unaffiliated/lupine-85/x-7392152) |
| 00:34.46 | lupine_85 | hihi. So i've just got an N900 and I'm setting up a very basic SIP-only (for now) asterisk server to handle the calls. I've got two server-local accounts working, and they can call each other - next step is to enable ad-hoc calls to any fully-qualified sip address, but I'm not entirely sure how to go about it |
| 00:35.31 | lupine_85 | I understand extensions.conf can have macros, so presumably it's something like exten => (match-param),1,Dial(SIP/matched-param) |
| 00:35.45 | lupine_85 | just having a bit of trouble understanding the syntax |
| 00:36.01 | jaytee | the book might help |
| 00:36.13 | bmoraca_work | the book would definitely help |
| 00:36.13 | voipmonk | just a little |
| 00:36.14 | lupine_85 | mm, it's sat on the shelf at work :) |
| 00:36.15 | voipmonk | :) |
| 00:36.24 | bmoraca_work | ~book |
| 00:36.24 | 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 |
| 00:36.29 | voipmonk | u can download it |
| 00:36.35 | bmoraca_work | it's on the intarwebs |
| 00:36.36 | lupine_85 | ooh, rock on |
| 00:36.53 | bmoraca_work | damnit, my IRC client won't let me mispell the word "the"... t-e-h |
| 00:37.06 | lupine_85 | presumavbly there aren't so many changes between 1.4 and 1.6 |
| 00:37.42 | bmoraca_work | lupine_85, on the basics like that, no...not really. though, they did go back to "," instead of "|" as the option delimiter for functions and applications in the dialplan |
| 00:37.46 | bmoraca_work | that's the biggest one, though |
| 00:38.06 | *** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com) |
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| 00:39.06 | bmoraca_work | uhg, it always freaks me out...the qualify messages from Asterisk show up in my "debug voice dialpeer" on my AS5400 so i always see these "failed calls" go through |
| 00:39.26 | lupine_85 | coolio. thanks for the link. I am -> bed now so I'll have to get it working in the morning... *yawn* |
| 00:41.10 | bmoraca_work | goes home...finally |
| 00:41.40 | carrar | bed? It's on;y 4:41PM |
| 00:42.30 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 00:45.13 | Kobaz | yeah it's nap time |
| 00:45.30 | hardwire | napnapnap |
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| 01:01.39 | *** join/#asterisk madsara (i=madsara@2001:328:2002:f159:0:0:0:1) |
| 01:01.47 | madsara | So... I'm using iwar + iax2.... iaxclient seems to want to go to s@context-name, instead of _NXXXXXX@context-name.... so, what am I doing wrong? |
| 01:01.53 | madsara | I've tried using ${ARG1} or ${ARG2} to capture the number being dialed, to no avail. |
| 01:01.57 | madsara | Also tried ${CALLERID(dnid)} |
| 01:02.42 | *** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) |
| 01:02.56 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 01:02.59 | jaytee | muy, muy grande cojones!!! http://de.acidcow.com/pics/20100118/acid_picdump_130.jpg |
| 01:04.07 | Qwell | jaytee: http://www.youtube.com/watch?v=lDBrdl2sZWs |
| 01:04.23 | Qwell | far far better |
| 01:04.59 | p3nguin | Is there an appliance that integrates the media gateway and the DSLAM together? |
| 01:05.06 | *** join/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
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| 01:24.47 | *** part/#asterisk nny (n=scott@64.203.239.83) |
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| 01:32.39 | *** part/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
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| 01:37.44 | riddlebox | are the yum repos on the digium site able to used on centos? |
| 01:41.45 | riddlebox | nm I should have read more |
| 01:44.07 | *** join/#asterisk dennis00 (n=dennis@unaffiliated/dennis00) |
| 01:45.05 | dennis00 | Hi, some way when I start my Softphone it is always already logged in into Voipbuster, but needs 5 seconds to log in onto my Asterisk server. Is the cause obvious? |
| 01:48.15 | paulc | p3nguin: ZyXel IES something or other? a guy was playing with one the other day, seems to do what you want (ADSL on top of a VoIP provided FXO interface?) |
| 01:49.14 | *** part/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
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| 02:02.41 | batphone | anyone doing wholesale in here? |
| 02:03.35 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 02:03.52 | batphone | im getting a carrier message "no route found" downstream over three different providers in the northeast |
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| 02:24.10 | Raden | batphone, yea whats up ? |
| 02:24.17 | Raden | northeast where ? |
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| 02:43.17 | darkdrgn2k | any one know where i can find info for provisioning PAP2T-NA |
| 02:46.19 | *** join/#asterisk plut0 (n=cory@cpe-72-224-58-127.nycap.res.rr.com) |
| 02:47.01 | plut0 | does ReceiveFax work at all? keeps throwing "bad magic number" errors |
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| 03:07.47 | dlynes | plut0, how are you using it? |
| 03:08.10 | dlynes | darkdrgn2k, try broadbandreports.com, linksys/sipura forum |
| 03:08.28 | dlynes | darkdrgn2k, or maybe it was voxilla.org, linksys/sipura forum |
| 03:08.38 | plut0 | dlynes: myfax.com sending to my asterisk box over SIP |
| 03:08.49 | dlynes | plut0, no t.38? |
| 03:09.01 | dlynes | plut0, just straight ulaw? |
| 03:10.52 | plut0 | t38pt_udptl = yess |
| 03:11.10 | dlynes | plut0, have you checked to see if it's actually doing t.38? |
| 03:11.16 | plut0 | i have no idea |
| 03:11.22 | plut0 | do i need a certain codec enabled? |
| 03:11.29 | dlynes | plut0, and also, it's got a value of 'yess', not 'yes'? |
| 03:11.50 | plut0 | yes |
| 03:11.53 | plut0 | yess? |
| 03:12.02 | dlynes | well, that could be why it's not working |
| 03:12.08 | dlynes | ' yess' is not a valid value |
| 03:12.18 | plut0 | t38pt_udptl = yes |
| 03:12.22 | dlynes | ok |
| 03:12.32 | dlynes | so you typoed, instead of copying/pasting then...ok |
| 03:13.05 | plut0 | so what am i missing? |
| 03:13.15 | dlynes | which fax solution are you using? app_fax.so with spandsp, or spandsp with res_fax.so and res_digium_fax.so? |
| 03:13.35 | plut0 | i'm using ReceiveFax |
| 03:13.43 | dlynes | that doesn't tell me which one you're using |
| 03:13.49 | dlynes | they're both ReceiveFax |
| 03:14.44 | plut0 | good question |
| 03:15.08 | dlynes | Which version of asterisk are you using? |
| 03:15.21 | plut0 | 1.6.2.0 |
| 03:15.36 | dlynes | ok, and did you download fax for asterisk from digium's website? |
| 03:16.12 | plut0 | installed via gentoo |
| 03:16.21 | dlynes | so, that's a no, then |
| 03:16.23 | *** join/#asterisk pawz_ (n=pawz@ppp118-208-182-24.lns20.bne4.internode.on.net) |
| 03:16.24 | dlynes | right? |
| 03:16.36 | dlynes | and gentoo installed it via source, or binary? |
| 03:16.41 | plut0 | source |
| 03:16.43 | dlynes | ok |
| 03:16.48 | dlynes | one sec |
| 03:16.52 | plut0 | spandsp |
| 03:17.23 | dlynes | plut0, and you're trying to receive, or send? I can't remember which it was you were doing |
| 03:17.32 | plut0 | receive |
| 03:17.36 | plut0 | app_fax.so |
| 03:17.56 | dlynes | plut0, ok, after you finish receiving the fax, check the value of 'FAXMODE' |
| 03:18.18 | dlynes | plut0, if the convo went t.38, the value will be 'T38'; else it will be 'audio' |
| 03:18.35 | plut0 | well it crashed asterisk so thats pretty difficult |
| 03:18.43 | dlynes | plut0, type core show application receivefax |
| 03:18.45 | dlynes | oh |
| 03:19.01 | dlynes | you said you got an error...you didn't say it crashed asterisk |
| 03:19.26 | plut0 | well it didn't crash but it hung indefinitely |
| 03:19.34 | plut0 | and i had to kill the process and restart |
| 03:20.25 | dlynes | plut0, you know you can do 'udptl debug ip blahblahblah'? |
| 03:20.37 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 03:21.50 | dlynes | plut0, you can also do 'sip set debug ip blahblahblah' to see if the other end allows your end to negotiate t.38 |
| 03:22.04 | dlynes | plut0, also...one other quick question |
| 03:22.17 | dlynes | plut0, is the codec for this call set to anything other than alaw or ulaw? |
| 03:22.19 | *** join/#asterisk Mango (n=Mango@d154-20-97-118.bchsia.telus.net) |
| 03:23.27 | Mango | I have a string like one-two-three-four and I need to remove everything after the last - |
| 03:23.38 | Mango | I found the CUT function but it seems as if I need to know how many elements and I do not. |
| 03:23.45 | Mango | Can anyone suggest anything? |
| 03:24.00 | *** join/#asterisk jhirley (n=jhirley@adsl-161-132-103.mia.bellsouth.net) |
| 03:24.07 | dlynes | Mango, system(sed -e 's/..../g') |
| 03:24.36 | dlynes | Mango, erm...actually...that won't work |
| 03:24.42 | Mango | hehe |
| 03:25.18 | dlynes | Mango, curl(/path/to/my/php-url-that-does-a-regex-transformation-on?value=${MYVALUETHATIWANTOTRANSFORM}) |
| 03:25.33 | Mango | Heh |
| 03:25.40 | dlynes | Mango, wouldn't that work? |
| 03:25.46 | Mango | I do suppose it might. |
| 03:25.54 | Mango | I was looking for something that the server could do without php though :P |
| 03:26.00 | dlynes | Mango, or a perl cgi script, or whatever |
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| 03:30.07 | plut0 | dlynes: codec is set to ulaw |
| 03:31.29 | Mango | think I got it |
| 03:31.30 | Mango | echo abc-def-ghi-jkl | sed 's/-[^-]*$//g' |
| 03:32.13 | plut0 | dlynes: what else am i missing? |
| 03:32.35 | Mango | thanks for the typ dlynes |
| 03:32.38 | Mango | *tip |
| 03:34.53 | Mango | Hmm, would backticks be the best for capturing the output of a command? |
| 03:36.16 | dlynes | Mango, you're just trying to remove the final '-'? |
| 03:36.25 | Mango | and everything after it |
| 03:36.36 | dlynes | one sec |
| 03:37.37 | dlynes | Mango, echo "abc-def-ghi-jkl" | sed -e 's/^.*-\(.*\)/-\1/g' |
| 03:37.50 | Mango | right, but now I need to get that into Asterisk |
| 03:39.31 | dlynes | Mango, the system command stores the result in the '${SYSTEMSTATUS}' variable |
| 03:39.41 | dlynes | erm system dialplan application, rather |
| 03:41.00 | dlynes | Mango, backticks won't do anything...asterisk is an application, not a shell script |
| 03:41.32 | dlynes | plut0, ok, assuming the other end has t.38, it should negotiate it |
| 03:42.01 | dlynes | plut0, so, I would first start looking at your sip debug to see if both ends are trying to negotiate t.38 (it should appear as a sip reinvite to use udptl) |
| 03:42.27 | dlynes | plut0, if that doesn't happen, the other end probably doesn't support t.38, or your end didn't detect a fax tone |
| 03:42.42 | dlynes | plut0, if it does happen, then you need to look further by checking out your udptl debug |
| 03:43.01 | ChannelZ | Is there a problem with using the REGEX() function? |
| 03:43.15 | Mango | dlynes, ${SYSTEMSTATUS} always contains SUCCESS |
| 03:43.21 | ChannelZ | oh nm it's a true/false matcher |
| 03:43.54 | dlynes | Mango, joy |
| 03:45.06 | dlynes | Mango, yeah...I think your best option is curl |
| 03:45.26 | dlynes | Mango, in recent versions of asterisk, it's implemented as a function, rather than as an application |
| 03:45.46 | Mango | What's the difference? |
| 03:45.59 | dlynes | Mango, no difference, other than to asterisk |
| 03:46.13 | dlynes | Mango, to you, it's just a difference in how you use it, not how it works |
| 03:46.20 | Mango | I'm SURE there must be a better option than curl |
| 03:46.34 | Mango | And why is there a distinction between functions and applications if they're the same? |
| 03:46.36 | dlynes | Mango, you could write your own function :) |
| 03:46.48 | dlynes | Mango, functions act like variables |
| 03:46.57 | Mango | ah. |
| 03:47.08 | dlynes | Mango, if you need more functionality than that..especially more processing, applications are more what you're looking for |
| 03:47.26 | dlynes | Mango, applications can set multiple variables |
| 03:47.59 | Mango | I see. |
| 03:48.04 | dlynes | Mango, applications and functions can both return values, but using different methods....functions act as a return value; applications need to return their values in a dialplan variable |
| 03:48.17 | dlynes | Mango, understand? |
| 03:48.59 | dlynes | you know what they shoudl have, though? |
| 03:49.02 | dlynes | a sed function :) |
| 03:49.03 | Mango | I do. Thank you :) |
| 03:49.06 | Mango | hehe |
| 03:49.16 | Mango | or a way of getting the output from the one that's already there. |
| 03:49.20 | Mango | maybe I should learn ASGI |
| 03:49.21 | Mango | *AGI |
| 03:49.31 | random_mike | clear |
| 03:49.39 | random_mike | erm - wrong window |
| 03:49.41 | random_mike | carry on.... |
| 03:49.44 | Mango | Charge to 300! |
| 03:49.45 | Mango | Clear! |
| 03:49.54 | random_mike | l) |
| 04:19.55 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 04:19.55 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #freepbx #switchvox #asterisk-bugs |
| 04:20.16 | p3nguin | My current problem is iaxmodem isn't very good. |
| 04:20.24 | dlynes | p3nguin, no...it sucks horribly |
| 04:20.47 | p3nguin | I have received two of probably 30 faxes. |
| 04:20.54 | dlynes | p3nguin, i'm working on a solution to still use hylafax for the hylafax printer |
| 04:21.22 | dlynes | p3nguin, and then have that hylafax printer dump the fax spool, which i then read with a shell script and send it off via call files for asterisk |
| 04:21.24 | p3nguin | What format will app_fax leave the received faxes in? |
| 04:21.35 | dlynes | p3nguin, it only understand tiff |
| 04:21.46 | dlynes | p3nguin, digium's driver on the other hand understands both tiff and pdf |
| 04:22.12 | dlynes | p3nguin, that being said, you can use ghostscript to convert it to pdf and vice versa |
| 04:22.23 | dlynes | tiff2ps for tiff to pdf |
| 04:22.24 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 04:22.24 | *** mode/#asterisk [+o file] by ChanServ |
| 04:22.30 | dlynes | and ghostscript for the other direction |
| 04:22.33 | *** join/#asterisk seanjohn (n=john@static-173-50-101-14.nrflva.east.verizon.net) |
| 04:23.04 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 04:23.33 | dlynes | Znuff, can i get fax over my voip? |
| 04:23.44 | Znuff | stabs dlynes in the face |
| 04:23.48 | dlynes | lol |
| 04:23.51 | seanjohn | is there any way to specify options for disa like if they dial just "1", can a dial plan be specified? |
| 04:23.59 | seanjohn | yes, lynes |
| 04:24.19 | p3nguin | seanjohn: of course |
| 04:24.20 | seanjohn | http://asterfax.sourceforge.net/ |
| 04:24.31 | dlynes | seanjohn, disa allows you to specify a dialplan (dialplan context) |
| 04:24.49 | dlynes | seanjohn, whatever you've got in that dialplan context is whatever they can dial |
| 04:24.50 | p3nguin | Put in the context, and in that context, have an exten => 1,1,stuff() |
| 04:25.07 | dlynes | seanjohn, i was joking about the fax...i know it can be done |
| 04:25.25 | seanjohn | yes, penguin, but creating a regular dial plan within the same context causes another disa to run over top of the other one when transferring back to the disa |
| 04:25.27 | dlynes | seanjohn, znuff said earlier he felt like stabbing someone in the face that asks about fax over voip |
| 04:25.32 | seanjohn | goto(disa,s,1) |
| 04:25.43 | p3nguin | seanjohn: You have to create a new context for your disa. |
| 04:27.35 | dlynes | Disa(xxx-passcode,dialplancontext,callerid-that-you-want-to-masquerade-as,,p) |
| 04:27.41 | seanjohn | for each one ran after the next? |
| 04:27.59 | seanjohn | that would be crazy; i would have to make asterisk generate a sequential number and use that as a variable |
| 04:28.19 | dlynes | seanjohn, huh? |
| 04:28.42 | dlynes | seanjohn, sequential number for what? |
| 04:29.01 | seanjohn | ok, i have a menu before the disa and they can dial c-h-a-n-g-e to change their password; after they confirm it, it sets it but the next line is goto(disa,s,1) |
| 04:29.17 | seanjohn | when it goto's, it creates another disa on top of the original one |
| 04:29.24 | seanjohn | and it ignores input |
| 04:29.35 | seanjohn | try it |
| 04:29.49 | dlynes | seanjohn, ok, so 'disa' is just a dialplan context? you're not using the disa dialplan application? |
| 04:30.10 | seanjohn | but my disa is a dialplan with a strict set of rules user-based |
| 04:30.20 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-hsgzfvpuvesmdeiq) |
| 04:30.28 | seanjohn | at the end of setting them, it actually does disa(no-password) |
| 04:30.33 | dlynes | seanjohn, can you pastebin your dialplan so that we have a clue what you're talking about? |
| 04:31.12 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
| 04:32.14 | seanjohn | hold on |
| 04:38.18 | *** join/#asterisk e4voip (n=chatzill@96-42-138-136.dhcp.trcy.mi.charter.com) |
| 04:45.21 | dlynes | held on for too long...heading to bed now... |
| 04:46.34 | *** join/#asterisk Tim_Toady (n=moi@77.49.233.62.dsl.dyn.forthnet.gr) |
| 04:51.21 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 04:51.21 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #freepbx #switchvox #asterisk-bugs |
| 04:52.14 | *** join/#asterisk lordmortis (n=lordmort@124-169-197-222.dyn.iinet.net.au) |
| 04:55.28 | seanjohn | sorry, talking to 3 other people, phone text and internet |
| 05:00.42 | random_mike | utils.c: write() returned error: Broken pipe - hmmm something's not working write! |
| 05:05.31 | seanjohn | http://pastebin.ca/1757040 |
| 05:06.18 | seanjohn | the extension code is 242643 |
| 05:06.40 | seanjohn | to change their cid |
| 05:06.48 | *** join/#asterisk ChannelZ (i=channelz@burner.com) |
| 05:06.53 | seanjohn | the default cid is the cid they're coming from |
| 05:09.28 | dennis00 | Can somebody help me and find the problem in this call? http://pastebin.ca/1757044 |
| 05:10.23 | seanjohn | dennis, an output of /var/log/asterisk/full after placing the call would help better |
| 05:11.23 | *** join/#asterisk aceking5 (n=phelix@71-94-132-102.static.mtpk.ca.charter.com) |
| 05:11.36 | dennis00 | [Jan 19 06:07:47] NOTICE[25565] chan_sip.c: Call from '508' to extension '01071428xx' rejected because extension not found. |
| 05:12.19 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 05:12.29 | seanjohn | are you calling 0107142801 0107142802 or something similiar |
| 05:12.41 | *** join/#asterisk Tim_Toady (n=moi@77.49.189.186.dsl.dyn.forthnet.gr) |
| 05:12.47 | dennis00 | I am calling a IPkall number |
| 05:12.51 | dennis00 | I filled out that number with IPKall |
| 05:12.54 | dennis00 | that' s my budgetphone number |
| 05:13.01 | dennis00 | But I think I should not fill out that number there. |
| 05:13.08 | dennis00 | exten => 508,1,Goto(your-main-menu|s|1) Maybe I should fill in 508. |
| 05:13.20 | seanjohn | do you have the sip context as the context of your dialplan |
| 05:14.17 | Maximo | good nite all !!! |
| 05:14.41 | dennis00 | seanbright: Can I reuse names? |
| 05:16.37 | [TK]D-Fender | dennis00: its failing to match the number you TOLD THEM TO DIAL |
| 05:16.57 | dennis00 | It now says: "Your account is being called at the moment, therefore this call is being disconected now". |
| 05:17.19 | dennis00 | I have been able to fix it, but what i dont get is. |
| 05:17.34 | dennis00 | Sometimes exten first parameter refers to an extension and sometimes an actual phone number. |
| 05:18.27 | seanjohn | its how its organized |
| 05:18.28 | [TK]D-Fender | [00:11]<dennis00>[Jan 19 06:07:47] NOTICE[25565] chan_sip.c: Call from '508' to extension '01071428xx' rejected <--- go look in SIP DEBUG to see what PEER its matching and in what CONTEXT it is looking for '01071428xx' |
| 05:19.02 | dennis00 | this is in my context => exten => 508,1,Goto(inbound|s|1) |
| 05:19.57 | dennis00 | I'll try to use code I know |
| 05:20.01 | [TK]D-Fender | dennis00: it is looking for '01071428xx' <--- WAKE UP |
| 05:20.08 | dennis00 | and use playback to test. |
| 05:20.11 | dennis00 | I know!! |
| 05:20.16 | dennis00 | I have that in extensions.conf |
| 05:20.25 | [TK]D-Fender | dennis00: WRONG CONTEXT. |
| 05:20.39 | [TK]D-Fender | dennis00: go look at SIP DEBUG like I told you |
| 05:20.48 | dennis00 | sip.conf for extension 508 has context=010.... and extensions.conf [010...]. |
| 05:20.48 | dennis00 | OK |
| 05:21.30 | seanjohn | exten => s,1,Answer exten s,2,Playback(say-something) exten=> s,3,Hangup............you can use letters a through z to set a dialplan but x,i,h are reserved. Using a dialpattern or numbers catches numbers dialed |
| 05:21.54 | seanjohn | s is the part of the dial plan, s,1 is the order see 1,2,3 |
| 05:22.11 | [TK]D-Fender | seanjohn: "s" is worthless here |
| 05:22.20 | seanjohn | i use it a lot |
| 05:22.41 | dennis00 | It is working now. |
| 05:23.31 | dennis00 | I have another question, I want to forward a call to another dialplan. Should I use a goto? http://pastebin.ca/1757056 |
| 05:25.33 | [TK]D-Fender | dennis00: that isn't "forwarding". That is simply moving around your dialplan. |
| 05:25.49 | [TK]D-Fender | dennis00: and right now, and from now on, stop using "|" as a parameter delimiter in your dialplan |
| 05:26.44 | dennis00 | I copied that from http://www.voip-info.org/wiki/view/IPKall. |
| 05:26.58 | [TK]D-Fender | dennis00: WIKI is for of ancient garbage |
| 05:27.11 | [TK]D-Fender | full of* |
| 05:28.46 | dennis00 | but do you see why 1) works and 2) did not work? the first exten with playback worked. |
| 05:29.00 | dennis00 | also, how would you recommend setting 508? not by repeating exactly the same code right? |
| 05:29.22 | random_mike | can anyone advise if it is possible to turn up the debug on AGI scripts? I have a perl script that errors when trying to Say Alpha on CallerIDName - it generates this error "utils.c: write() returned error: Broken pipe" |
| 05:29.27 | [TK]D-Fender | dennis00: You seem to have a problem listening.... |
| 05:29.52 | [TK]D-Fender | [00:20]<[TK]D-Fender>dennis00: it is looking for '01071428xx' <--- WAKE UP <--- this was the EXTENSION it was looking for. Get the FUCK OUT OF SIP.CONF |
| 05:29.58 | [TK]D-Fender | :p |
| 05:30.11 | random_mike | *walks off quietly* |
| 05:30.11 | dennis00 | I changed the context=. |
| 05:30.27 | dennis00 | context=ipkallinbound now... |
| 05:30.44 | [TK]D-Fender | dennis00: And you didn't even show what PEER it matched. You are taking random actions, not showing the RESULT and then asking what to do. |
| 05:30.57 | [TK]D-Fender | dennis00: How do *I* know you put that in the right place? |
| 05:31.16 | [TK]D-Fender | dennis00: You are showing a complete lack of communication and debugging skills |
| 05:34.29 | dennis00 | well, it works now. |
| 05:48.50 | *** join/#asterisk jakent (n=john@c-98-233-13-157.hsd1.va.comcast.net) |
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| 05:50.39 | *** part/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
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| 06:37.53 | random_mike | is anyone able to assist with some perl agi issues I'm experiencing? |
| 06:38.31 | random_mike | namely the error "utils.c: write() returned error: Broken pipe" when trying to say_phonetic or say_alpha ? |
| 06:39.50 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 06:41.09 | random_mike | matter of fact it looks like it only happens when it tries to read out the callerID Name |
| 06:41.17 | random_mike | which is set in both examples. |
| 06:50.34 | *** join/#asterisk |Cybex| (n=John@atwork-21.r-212.178.82.atwork.nl) |
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| 07:06.42 | Znuff | wow |
| 07:10.21 | *** join/#asterisk ZX81 (n=Matt_Rid@121-74-230-221.telstraclear.net) |
| 07:10.38 | *** part/#asterisk ZX81 (n=Matt_Rid@121-74-230-221.telstraclear.net) |
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| 07:11.33 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-cecyksnjckwkjmkc) |
| 07:14.09 | aceking5 | anyone here ever used flowroute? |
| 07:14.30 | drmessano^ | I have |
| 07:15.13 | aceking5 | when I use sipx, their server fails to send me a session progress after the 100 trying |
| 07:15.16 | aceking5 | so i get no ring |
| 07:15.25 | aceking5 | it works with asterisk and x-lite though |
| 07:15.34 | aceking5 | i just can't put my finger on the problem |
| 07:16.48 | drmessano^ | Dunno... You need more than Asterisk? :P |
| 07:17.22 | aceking5 | yeah i just like the provisioning features sipx has |
| 07:17.30 | aceking5 | i thought of using asterisk as a gateway for sipx |
| 07:17.37 | aceking5 | but idk how caller id would pass through |
| 07:19.01 | *** join/#asterisk mpe (n=mpe@gate.ipvision.dk) |
| 07:19.27 | aceking5 | can asterisk take the From header from a call coming from sipx and pass it on as a From header in the sip invite to my itsp? |
| 07:22.46 | *** join/#asterisk klochan (n=klochan@213.152.137.43) |
| 07:23.53 | *** join/#asterisk mlarsen (n=mlarsen@212.37.141.188) |
| 07:24.23 | drmessano^ | It can. |
| 07:24.55 | aceking5 | how? |
| 07:26.26 | drmessano^ | http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+sendrpid |
| 07:27.11 | *** join/#asterisk sun28 (n=light@sun28.ipfw.su) |
| 07:27.21 | aceking5 | nice ill look into that |
| 07:27.22 | aceking5 | thanks |
| 07:28.28 | sun28 | moin \o/ |
| 07:35.54 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:44.20 | *** join/#asterisk spenguin[work] (n=penguin@122.182.0.38) |
| 07:50.01 | *** join/#asterisk infobot (i=ibot@rikers.org) |
| 07:50.01 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.2.1 (2010/01/15), Asterisk 1.6.1.13 (2010/01/15), 1.6.0.21 (2010/01/15), 1.4.29 (2010/01/15), *-Addons 1.6.2.0 (2009/12/18), 1.6.1.2 (2009/12/02), 1.6.0.4 (2009/12/02), 1.4.10 (2009/12/02), dahdi-linux 2.2.0.2 (2009/07/23), dahdi-tools 2.2.0 (2009/06/24), Libpri 1.4.10.2 (2009/10/20) -=- Related channels: #freepbx #switchvox #asterisk-bugs |
| 07:52.37 | spenguin[work] | hrm it seems like if I punch in too quick it doesnt do it right |
| 07:55.52 | spenguin[work] | one of the extentions is 11 |
| 07:56.09 | spenguin[work] | and if punched in too quick, asterisk takes it as 1 |
| 07:56.18 | spenguin[work] | what can I do in this case? |
| 07:58.31 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
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| 08:02.13 | ManxPower-work | spenguin[work]: how are the calls arriving to Asterisk? |
| 08:02.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 08:08.10 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
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| 08:22.45 | spenguin[work] | ManxPower-work: well an account with a voip provider forwards the calls |
| 08:23.17 | spenguin[work] | in this case I was testing the ivr by calling the number, which lands where Im calling from |
| 08:23.31 | spenguin[work] | my call is via sip |
| 08:23.43 | *** join/#asterisk Faustov (i=user@gentoo/user/faustov) |
| 08:25.24 | spenguin[work] | well normally id have 4 digit extentions |
| 08:25.34 | spenguin[work] | which cant really be punched fast enough |
| 08:25.59 | spenguin[work] | but some 2 digit extentions like 11, 22 etc seem to have trouble |
| 08:26.15 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 08:28.29 | ChannelZ | sounds like their DTMF detection is not so good |
| 08:29.42 | spenguin[work] | ok |
| 08:30.19 | spenguin[work] | since asterisk seems to see only one digit when that happens |
| 08:30.30 | spenguin[work] | I think it can be blamed on them |
| 08:30.30 | ChannelZ | at least I'm assuming you're getting dtmf out-of-band |
| 08:30.46 | spenguin[work] | is there a better method to confirm? |
| 08:31.10 | Amorsen | Get them to switch to dtmfmode=inband |
| 08:31.13 | Amorsen | Temporarily |
| 08:31.24 | Amorsen | Assuming that you're on Alaw or ulaw of course |
| 08:31.25 | ChannelZ | well not really, except for _you_ to do the dtmf reading if they are doing it badly |
| 08:31.32 | ChannelZ | what he said |
| 08:33.52 | aceking5 | i cant figure out how to pass the From: header into caller id whn using asterisk as a gateway for sipx :( |
| 08:34.07 | spenguin[work] | dtmfmode=auto |
| 08:34.13 | spenguin[work] | thats what weve got |
| 08:34.31 | Amorsen | I'd recommend setting it explicitly, in general |
| 08:34.36 | spenguin[work] | also seem to have relaxdtmf=yes |
| 08:34.46 | Amorsen | You definitely want relaxdtmf=no |
| 08:34.47 | spenguin[work] | in zapata.conf |
| 08:35.03 | Amorsen | Ah, I thought you meant sip.conf |
| 08:35.15 | spenguin[work] | nah |
| 08:35.43 | *** join/#asterisk shamelessn00b (n=chatzill@58-65-172-114.nayatel.pk) |
| 08:35.45 | Amorsen | zapata.conf doesn't have any effect if you don't have hardware cards |
| 08:35.50 | shamelessn00b | hi all |
| 08:35.57 | ChannelZ | wait, are you having these problems on SIP channels or Zap channels (analog?) |
| 08:36.08 | spenguin[work] | sip |
| 08:36.22 | spenguin[work] | so I shouldnt be bothered about zapata.conf then? |
| 08:36.31 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-lwypcrzllfmjplpg) |
| 08:36.32 | shamelessn00b | I want to set up a FastAGI server atm |
| 08:36.43 | ChannelZ | no |
| 08:36.48 | shamelessn00b | are there any solutions available that offer sca;labitity |
| 08:36.49 | shamelessn00b | ?? |
| 08:36.49 | spenguin[work] | but I do have dtmfmode=auto set in sip.conf |
| 08:37.02 | shamelessn00b | scalability* |
| 08:39.07 | Amorsen | spenguin[work]: You'll have to figure out how your providers sends DTMF and change dtmfmode to match |
| 08:39.26 | spenguin[work] | Amorsen: ok |
| 08:39.28 | spenguin[work] | will do |
| 08:40.28 | *** join/#asterisk benngard (n=benngard@213.88.138.230) |
| 08:42.31 | aceking5 | flowroute passes on a static ANI for every call? wtf? |
| 08:46.20 | *** join/#asterisk c0rnoTa (n=c0rnoTa@109.188.57.215) |
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| 08:49.35 | spenguin[work] | Amorsen: http://pastebin.com/m6f6c141d |
| 08:49.58 | spenguin[work] | so it seems like I could specify RFC2833 |
| 08:50.19 | spenguin[work] | but isnt Auto a better option |
| 08:50.20 | spenguin[work] | ? |
| 08:50.49 | Amorsen | You're relying on Asterisk to guess correctly |
| 08:51.29 | spenguin[work] | ok |
| 08:51.32 | Amorsen | Try dtmfmode=rfc2833 and relaxdtmf=no (in sip.conf) |
| 08:51.40 | spenguin[work] | ok |
| 08:51.44 | Amorsen | It's cheap to try... |
| 08:51.45 | TommyBotten | What is 'relaxdtmf' ? |
| 08:52.19 | *** join/#asterisk war9407 (i=war@liquidswords.org) |
| 08:52.27 | Amorsen | TommyBotten: A way for Asterisk to cope with non-standard-compliant peers, I believe |
| 08:52.33 | *** join/#asterisk niekie (i=quasselc@CAcert/Assurer/niekie) |
| 08:52.50 | Amorsen | Also a way for Asterisk to try to interpret inband DTMF even if you specify rfc2833. Grr. |
| 08:53.44 | Amorsen | You also want rfc2833compensate to be off, but you'd probably know it if it was on |
| 08:54.16 | imcdona | So Telus screwed up a remote call forward request. I call in and am told it will take 4 hours to fix. I call again in 4 hours to find it will take 24 hours to fix. I demand to speak with a supervisor. The supervisor says he will "call forward the lines while the techs are working on the problem". I tell him that was the reason I put in the trouble ticket. He tells me they don't call forward unless a customer specificly asks for a supervi |
| 08:55.00 | *** join/#asterisk stephank (n=traveler@212.178.158.35) |
| 08:56.22 | ChannelZ | "We don't help you unless you beg." |
| 08:57.29 | imcdona | no kidding |
| 08:58.01 | *** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net) |
| 08:59.05 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 08:59.12 | shamelessn00b | load balancing on FastAGI server any possible solutions available y/n?? |
| 09:01.20 | *** join/#asterisk Ceeriael (n=ceeriael@93.167.108.90) |
| 09:03.18 | Ceeriael | I have a strange problem: extention wich is not internal, I can register, and receive calls with sound. If i try and call the extention the phone will ring, i can pick up... but NO sound in ether end... It strange when it works when the extention is calling out.... |
| 09:03.23 | Ceeriael | then it works. |
| 09:03.39 | Ceeriael | The phone im using is a Zyxel prestige 2000w v2 phone. |
| 09:04.04 | TommyBotten | Ceeriael: Is there a NAT firewall involved? |
| 09:04.04 | Ceeriael | Any idea what could be the problem ? |
| 09:04.08 | imcdona | sounds like a NAT issues |
| 09:04.31 | Ceeriael | then why would it work when its calling out ? |
| 09:06.36 | ChannelZ | Each side requests what port the other should send the voice traffic to |
| 09:07.21 | imcdona | forward ports udp 10,000-20,000 and udp 5060 on your router to asterisk. |
| 09:07.29 | Ceeriael | have done that. |
| 09:07.40 | Ceeriael | its working fine withe the trunks i have connected to the asterisk |
| 09:07.42 | imcdona | have you set externip in sip.conf? |
| 09:07.53 | Ceeriael | i have 15 external trunks i use for access to PSTN |
| 09:08.03 | Ceeriael | yes. my external ip is in the config |
| 09:08.34 | imcdona | how about localnet is sip.conf? |
| 09:08.49 | Ceeriael | also there |
| 09:09.18 | Ceeriael | if i use an x-lite client insteed of the Zyxel phone it works.... |
| 09:09.34 | TommyBotten | is allowreinvite set to yes? |
| 09:09.35 | Ceeriael | but before reinstalling my asterisk the zyxel phone worked ! |
| 09:09.36 | TommyBotten | It should not |
| 09:09.57 | Ceeriael | canreinvite = no |
| 09:10.00 | imcdona | what codec is set on the zyxel? set it to disallow=all and allow only one codec on each phone |
| 09:10.39 | Ceeriael | what codec should i use ? (dont have any licenses ?) |
| 09:11.04 | imcdona | ulaw or alaw |
| 09:11.08 | Ceeriael | ok |
| 09:11.19 | Ceeriael | disallow = ALL ; allow = ulaw |
| 09:11.21 | Ceeriael | check =) |
| 09:11.32 | imcdona | make sure the zyxel is also configured to use just one codec as well |
| 09:11.44 | Ceeriael | yse. |
| 09:11.50 | imcdona | in the phone itself |
| 09:11.55 | Ceeriael | thx =) ill try that. be back to ether thank you or to cry :D |
| 09:13.02 | stephank | Downloads links on asterisk.org still point to 1.6.2.0 and 1.4.28. I may be a bit paranoid, but any reason why they don't point to 1.6.2.1 and 1.4.29 yet? |
| 09:13.59 | ChannelZ | more than likely just a cock-up |
| 09:14.16 | stephank | hehe |
| 09:14.37 | ChannelZ | There's more than a couple things wrong with the website |
| 09:15.09 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:15.38 | *** part/#asterisk Tech_Travis (n=Administ@cpe-76-168-191-127.socal.res.rr.com) |
| 09:15.42 | *** join/#asterisk dandre (n=daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 09:17.38 | Ceeriael | I KISS YOU !!! =) thx TommyBotten and imcdona :) |
| 09:17.52 | *** join/#asterisk ruied (n=ruied@bl9-254-142.dsl.telepac.pt) |
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| 09:33.25 | *** join/#asterisk mpe_ (n=mpe@gate.ipvision.dk) |
| 09:34.39 | aceking5 | are there any carriers that pass through incoming blocked cid's even on non-tollfree DIDs? |
| 09:35.20 | *** join/#asterisk mbrevda (n=mbrevda@unaffiliated/mbrevda) |
| 09:35.44 | mbrevda | tzafrir_laptop: mornin. mah zeh? line 0: Unable to open master device '/dev/dahdi/ctl' |
| 09:36.05 | tzafrir_laptop | Do you actually have /dev/dahdi/ctl ? |
| 09:36.13 | tzafrir_laptop | Where do you get that error? |
| 09:36.14 | ChannelZ | I don't think that's in anyone's hands, if someone blocks their CID it's probably done on their telco's side and never even sent..? |
| 09:36.22 | *** join/#asterisk mpe__ (n=mpe@gate.ipvision.dk) |
| 09:36.35 | mbrevda | tzafrir_laptop: nope :( I get the error on dahdi_cfg |
| 09:36.51 | ChannelZ | did you start the drivers? |
| 09:37.02 | mbrevda | any idea what package provides that? |
| 09:37.15 | ChannelZ | dahdi.. |
| 09:37.21 | ChannelZ | there's usually a startup script |
| 09:37.32 | ChannelZ | /etc/init.d/dahdi start or somesuch |
| 09:37.58 | *** join/#asterisk arekm (i=arekm@pld-linux/arekm) |
| 09:38.23 | arekm | hi, /me wonders which echo cancellation alg produces best result in most of common cases? MG2 or KB1? |
| 09:38.49 | aceking5 | ChannelZ, CID is always sent... calling party's telco sends CID along with a privacy flag and receicing telco receives CID with privacy flag but mostly doesn't pass on CID to the user |
| 09:38.52 | aceking5 | afaik |
| 09:39.13 | aceking5 | thats how flowroute passed CID's a year ago |
| 09:39.19 | aceking5 | and how trapcall.com works |
| 09:39.31 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
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| 11:16.04 | razu_ | have anyone configured seemless routing between 2 asterisk boxes ? so that in case one trunk goes down everything (including live calls) will be routed to another trunk ... |
| 11:18.48 | MAbbas | Hi All, |
| 11:18.54 | MAbbas | I am generating calls to asterisk using WinSip. Problem is, WinSip generated calls go to "default" context, I want calls to be handled by my "test" context. |
| 11:19.35 | MAbbas | Calls are generated in same LAN as asterisk .. |
| 11:20.04 | MAbbas | how to go about with changes in sip.conf? |
| 11:29.58 | *** join/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
| 11:39.25 | Delido1983 | hello, i need help by dialing out with asterisk. all is okay but i become an busy signal by outgoing calls outside from germany. Calls from / to germany are okay and the prefix is allright (+49201...) Is there an problem with the international prefix in chan-dahdi.conf? |
| 11:39.48 | *** join/#asterisk lost_sou1 (n=noymfb@cpe-74-71-234-100.twcny.res.rr.com) |
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| 11:59.12 | donatas | who can explain why this occurs if I take horn (not voip) and got BUSY ? http://p.defau.lt/?XhTGAggyF6vgziDX6sMVMQ |
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| 13:14.06 | *** join/#asterisk [TK]D-Fender (n=chatzill@216.191.106.163) |
| 13:18.14 | beek | morning [TK]D-Fender |
| 13:18.23 | [TK]D-Fender | beek: Mornin' |
| 13:19.56 | *** part/#asterisk Delido1983 (n=Delido19@212.144.236.196) |
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| 13:41.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:42.29 | leifmadsen | meckaleckahi mechakhineyho! |
| 13:42.36 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 13:43.54 | TommyBotten | ... .. and the same to you! ... ? |
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| 14:12.55 | Katty | morning |
| 14:13.42 | coppice | not for another 1 hour 47 minutes |
| 14:17.58 | *** join/#asterisk highvoltz (i=rogers@bling.bling.org) |
| 14:20.29 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
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| 14:26.05 | *** join/#asterisk RobH (n=robh@rob.tech.wikimedia.org) |
| 14:26.41 | beek | Morning Katty |
| 14:27.21 | creativx | sigh |
| 14:28.34 | *** join/#asterisk koollman (n=samson_t@ns301422.ovh.net) |
| 14:31.08 | *** join/#asterisk chuckf (n=chuckf@ubuntu/member/chuckf) |
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| 14:33.30 | *** join/#asterisk huey23 (n=pleasedu@65.111.241.212) |
| 14:33.33 | *** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:1c9c:7ab6:a077:acde) |
| 14:33.37 | cusco | hi |
| 14:34.11 | cusco | is there some kind of memory leak in 1.6? |
| 14:35.05 | cusco | sl |
| 14:35.27 | cusco | asterisk is using 1.4% of memory when I start it, and ech day it grows about 10% |
| 14:35.35 | cusco | right now it is using 21% |
| 14:35.46 | cusco | tomorrow it will be using about 30% |
| 14:37.03 | Katty | hi beek |
| 14:37.04 | huey23 | i cannot get asterisk to attach the voicemail audio to the e-mail message with asterisk 1.4.28, any thoughts : http://pastebin.com/m23db4f66 |
| 14:37.52 | Katty | only one squirrel out this morning. that's unusual. |
| 14:38.02 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 14:38.34 | cusco | not enough peanuts |
| 14:38.42 | Katty | there's plenty of peanuts. trust me. |
| 14:40.38 | beek | Katty, somewhere in your neighborhood someone is cooking squirrel pot pie. |
| 14:40.40 | cusco | I used to go every weekendo to this park.. take a bag of peanuts, and watch them squirrels breaking in the peanut's shell |
| 14:41.15 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 14:42.00 | Gido-E | cusco you know pmap? |
| 14:42.06 | *** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net) |
| 14:44.27 | Gido-E | cusco with pmap, you can maybe see what the problem is? |
| 14:45.53 | Katty | beek: i highly doubt it. |
| 14:46.23 | [TK]D-Fender | huey23: So far you don't have any VM boxes at all.... |
| 14:47.21 | cusco | Gido-E: network server for robots and sensors? |
| 14:47.35 | [TK]D-Fender | huey23: Or the rest of a sane configuration for e-mail |
| 14:49.05 | lordmortis | is there a way to answer a call that's in a queue from an extension that's not a member? |
| 14:49.19 | Naikrovek | ** extension |
| 14:49.20 | Naikrovek | i think |
| 14:49.34 | Gido-E | cusco ? |
| 14:49.36 | _abc_ | Katty: where do you count the squirrels ? Canada ? |
| 14:49.36 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 14:50.01 | cusco | Gido-E: thats the description of libpmap |
| 14:50.04 | cusco | lol |
| 14:50.14 | Gido-E | pmap - report memory map of a process |
| 14:50.27 | [TK]D-Fender | lordmortis: Depends on what kind of member is being rung, etc |
| 14:51.04 | Katty | _abc_: my front yard. |
| 14:51.09 | Katty | _abc_: my front yard is not in canada. |
| 14:51.21 | lordmortis | okay, i've got incoming calls, and certain extenaions need to ring, but anyone from any extension needs to be able to answer the call |
| 14:51.33 | *** join/#asterisk af_ (n=getsmart@88-149-240-85.dynamic.ngi.it) |
| 14:51.35 | [TK]D-Fender | lordmortis: And you haven't answered my question |
| 14:51.40 | _abc_ | Katty: seems to be in Jackson ... |
| 14:51.58 | Katty | _abc_: There mgiht be a jackson in canada, i don't know. |
| 14:52.13 | _abc_ | what's the deal on using nortel ip phones (new ones) with asterisk ? |
| 14:52.24 | Katty | what's the deal? |
| 14:52.27 | Naikrovek | heh |
| 14:52.29 | Naikrovek | what's what deal |
| 14:52.30 | _abc_ | Katty: Jackson, Missouri accourding to my geoip tool run against your email whois |
| 14:52.37 | lordmortis | [TK]D-Fender: it's a ringall queue? and it's not a member of the queue that i want to be able to answer the call (i could have them dial a number to become agents or something but i'd prefer to let them dial something to just answer) |
| 14:52.58 | Katty | Naikrovek: dude. what's YOUR deal. |
| 14:53.08 | _abc_ | well do they work, anyone seen doing the 'ballmer' trick after using them ? (running in circles pushing war cries) |
| 14:53.09 | Katty | Naikrovek: ;> |
| 14:53.17 | Naikrovek | Katty: even i don't know taht |
| 14:54.59 | lordmortis | nevermind i think i got it |
| 14:55.01 | [TK]D-Fender | lordmortis: And you haven't answered my question <---------- |
| 14:55.19 | [TK]D-Fender | lordmortis: depends on what member IS BEING RUNG. |
| 14:56.55 | Gido-E | [TK]D-Fender there is no bel ringing :-) |
| 14:57.19 | [TK]D-Fender | Gido-E: Who said it had to make sound? :) |
| 14:57.54 | *** join/#asterisk TimeRider (n=steve@nat73.mia.three.co.uk) |
| 14:59.01 | huey23 | [TK]D-Fender: i do have mailboxes set up, i was just unaware you needed the mailbox info http://pastebin.com/m5fde6275 |
| 15:00.32 | Naikrovek | http://www.flickr.com/photos/kratiu/4287730820/ |
| 15:00.33 | cusco | using pmap, I don't know what am I looking for... Gido-E http://pastebin.com/f5c14059a |
| 15:01.39 | [TK]D-Fender | huey23: So far you aren't using the right field. |
| 15:01.53 | [TK]D-Fender | huey23: 3rd is supposed to be your e-mail address |
| 15:02.20 | huey23 | ok, so drop the name from all the mailboxes |
| 15:02.26 | Gido-E | cusco check in 4 hours again. but it uses 1.3 gig now. |
| 15:02.30 | huey23 | this is an upgrade from an earlier version |
| 15:02.33 | [TK]D-Fender | huey23: No, you got the parameters wrong |
| 15:02.40 | [TK]D-Fender | huey23: don't drop, get it right |
| 15:02.44 | huey23 | kk |
| 15:02.49 | [TK]D-Fender | huey23: Doesn't look like you read the sample config |
| 15:02.51 | cusco | Gido-E: what is all that [anon] |
| 15:03.12 | Gido-E | cusco i think asterisk |
| 15:04.26 | cusco | :/ ok |
| 15:08.04 | *** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net) |
| 15:12.50 | highvoltz | running asteriskNOW - getting an error listed on a default install status after adding a user account... Failed to copy from module agi-bin ideas? |
| 15:13.13 | Katty | blahhhhh |
| 15:13.15 | ManxPower-work | ~freepbx |
| 15:13.16 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 15:13.18 | ManxPower-work | ~asterisknow |
| 15:13.19 | infobot | extra, extra, read all about it, asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons. Please seek support in #asterisknow instead. |
| 15:13.38 | highvoltz | thanks :) |
| 15:15.14 | Katty | falls asleep on keybbbbbbbbbbbbbbbbbbbbbbbbb |
| 15:16.14 | *** join/#asterisk nicknick (n=administ@host213-123-201-13.in-addr.btopenworld.com) |
| 15:16.54 | _abc_ | -> Katty PINGGGG |
| 15:18.04 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 15:18.04 | _abc_ | seriously, had anyone here got success/failure stories to share about nortel 11xy series phones with asterisk |
| 15:18.17 | _abc_ | give me the good the bad and the ugly |
| 15:18.27 | _abc_ | i am doing my own research |
| 15:18.35 | _abc_ | but 2nd opinions would be nice |
| 15:19.16 | Katty | _abc_: Yes? |
| 15:20.07 | _abc_ | just making sure you don't sleep too much >:~ |
| 15:20.10 | *** join/#asterisk beers218 (i=45c6b055@gateway/web/freenode/x-ugsetrfjqrewzuaq) |
| 15:20.13 | Katty | heh |
| 15:20.27 | _abc_ | it's morningish in the us now |
| 15:22.06 | *** join/#asterisk casix (n=casix@xenpbxedifici.adamvozip.es) |
| 15:22.09 | casix | hello |
| 15:23.00 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 15:23.31 | casix | how can I continue with a channel after a meetme? |
| 15:26.30 | beers218 | I have my PBX pointing a DID to a macro for handling faxes, and i can hear the tones when i dial the DID from my cell phone. when i dial the DID from the fax machine, no tones are played and the fax machine doesnt start sending the document. any sugguestions? |
| 15:26.35 | Kobaz | casix: core show application meetme |
| 15:26.40 | [TK]D-Fender | casix: have more priorities |
| 15:27.17 | Bladerunner05 | if an incoming call is assigned to user 100 and it didn't answer how may I get that call to answer ? |
| 15:27.27 | *** join/#asterisk utahsaint (n=utahsain@mail.ntegratedsolutions.com) |
| 15:27.39 | ManxPower-work | Bladerunner05: your question makes no sense |
| 15:28.50 | Bladerunner05 | in a context like press 1 to sales (user1,user2) if user1 & user2 can't answer the call and I'm user3 how may I get that call ? |
| 15:29.43 | [TK]D-Fender | Bladerunner05: Add another priority to dial your phone after theirs, or go read "core show application pickup" |
| 15:29.45 | ManxPower-work | Bladerunner05: um, dial that call after you try calling the other users. |
| 15:30.24 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:31.14 | Bladerunner05 | ManxPower-work: I don't understand, please axplain me if u can |
| 15:31.33 | ManxPower-work | Bladerunner05: Are you running a GUI for Asterisk? |
| 15:32.45 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 15:32.45 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:32.51 | ManxPower-work | I thought so. |
| 15:33.11 | Katty | wtb pillow. |
| 15:34.30 | koollman | Hi. I'm looking for a specific functionnality, but can't even find its name. How would you call a phone that displays current 'busy' lines and such, used by a human phone operator (the person that receive the calls first and can redirect them, with a huge phone and many blinky leds ? :) ) |
| 15:34.41 | dlynes | beers218, i'd say you have faulty audio reproduction on the fax machine, then |
| 15:35.41 | ManxPower-work | koollman: BLF aka Busy Lamp Field |
| 15:36.04 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 15:36.16 | beers218 | dlynes: so your saying my fax speaker is not working properly? |
| 15:36.50 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 15:36.56 | koollman | ManxPower-work: thanks! that's incredibly hard to find :) |
| 15:37.40 | ManxPower-work | koollman: There is something similar called SLA aka Shared Line Appearances. Asterisk does not really support SLA. It does support BLF with supported phones. |
| 15:37.44 | *** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:37.44 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:37.47 | Bladerunner05 | ManxPower-work: is better to user gui ? |
| 15:37.56 | koollman | ok. now I will 'just' need such a phone. Thank you very much |
| 15:37.59 | ManxPower-work | Bladerunner05: We do not support GUIs here. |
| 15:38.13 | ManxPower-work | koollman: Polycom 650 w/Sidecar |
| 15:38.32 | ManxPower-work | Bladerunner05: put a copy of your option 1 dialplan on pastebin.ca |
| 15:38.50 | dlynes | beers218, who knows? I don't know what's wrong with your end, because i'm not there |
| 15:39.17 | ManxPower-work | beers218: How, exactly, is the fax machine connected to Asterisk? |
| 15:39.36 | dlynes | beers218, but if you hear the beeping on the cell phone and you don't hear it on the fax machine, and they're both taking exactly the same point of entry, using the same technology, then obviously there's something wrong with the audio reproduction on the fax machine |
| 15:40.00 | *** join/#asterisk Akiraa (n=Akiraaaa@79.112.27.47) |
| 15:40.55 | *** join/#asterisk minotaur01 (n=minotaur@24.215.3.50) |
| 15:40.57 | Bladerunner05 | ManxPower-work: http://www.pastebin.ca/1757549 |
| 15:41.14 | beers218 | well the fax machine isnt connected to our PBX. its connected to our IAD, as an analog line. im attempting to fax from my fax into asterisk, and have it receive the fax to email |
| 15:41.19 | dlynes | beers218, if they're not using the same point of entry, using the same technology, it's probably an asterisk configuration issue |
| 15:41.36 | ManxPower-work | Bladerunner05: http://pastebin.ca/1757550 |
| 15:42.20 | dlynes | beers218, so, is it: fax->analog line->phone company's pstn network->analog line->asterisk->email? |
| 15:42.27 | Bladerunner05 | ManxPower-work: so if user1 and user2 didn't answer it ring to user3 |
| 15:42.36 | ManxPower-work | Bladerunner05: correct. |
| 15:42.50 | [TK]D-Fender | Bladerunner05: Witha giant assumption about what your macro even does |
| 15:42.53 | dlynes | beers218, and is it also: cell phone->phone company's pstn network->analog line->asterisk? |
| 15:42.54 | beers218 | dlynes: i hate to say it but its VOIP |
| 15:42.57 | Bladerunner05 | ManxPower-work: but if I have more than 3 user and one free hear the call how can respond ? |
| 15:43.06 | [TK]D-Fender | Bladerunner05: PICKUP |
| 15:43.07 | dlynes | beers218, ulaw, or udptl(t.38)? |
| 15:43.15 | beers218 | dlynes: ulaw |
| 15:43.19 | beers218 | at this time |
| 15:43.19 | ManxPower-work | Bladerunner05: I answered your question. |
| 15:43.21 | dlynes | beers218, that's why it's not working |
| 15:43.27 | Bladerunner05 | <[TK]D-Fender>: is easy to setup ? |
| 15:43.30 | dlynes | ~fax |
| 15:43.30 | infobot | fax is probably The honor of designing the first fax *service* in actual use goes to Giovanni Caselli, an Italian abbot, born in Siena in 1815, who turned his hand to science and was, by 1849, editing a scientific magazine. In 1856 he claimed that he had developed a device, which he called a "pantelegraph," that could send facsimiles of images and text. Napoleon III did not come up with the idea, he merely backed it. |
| 15:43.38 | dlynes | ~foip |
| 15:43.39 | infobot | rumour has it, foip is Fax over IP. This requires funtionality standardised by T.38, realtime fax over IP, or T.37, store-and-forward via email. See http://soft-switch.org/foip.html for more detailed info about the subject |
| 15:43.52 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 15:43.56 | dlynes | hrm |
| 15:44.20 | [TK]D-Fender | Bladerunner05: Go read the instructions |
| 15:44.20 | dlynes | i know infobot's got something derogatory about fax over ip...I just don't know what the key phrase is |
| 15:44.26 | ManxPower-work | I still don't see how Asterisk is supposed to receive faxes. |
| 15:44.54 | Gido-E | i use voip and fax in and outcoming. Works perfect, i only use ulaw. |
| 15:45.04 | dlynes | ManxPower-work, it's actually fax machine->analog line->pstn network->itsp->sip (ulaw)->asterisk->email |
| 15:45.25 | ManxPower-work | dlynes: you don't magically turn sip into a fax. He is missing something |
| 15:45.29 | dlynes | Gido-E, congratulations...you're blessed and lead a charmed life |
| 15:45.52 | *** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com) |
| 15:45.56 | beers218 | i understand this, makes perfect sense... but when we get a fax from the PSTN->my telco/ISP->SIP->faxmachine it works fine all the time |
| 15:46.11 | dlynes | Gido-E, either that, or you're using a pri or an analog line in there that you're not mentioning, and then you're still leading a charmed life |
| 15:46.28 | Gido-E | dlynes sorry for my ubernes. |
| 15:46.31 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 15:46.33 | *** join/#asterisk tty (n=tty@77.49.232.215.dsl.dyn.forthnet.gr) |
| 15:46.57 | ManxPower-work | I discovered the secret to getting reliable faxing with Asterisk 8 years ago. |
| 15:47.14 | dlynes | ManxPower-work, analog bridging? |
| 15:47.17 | ManxPower-work | I discovered that if you want reliable faxing you will install a standard analog line from the telco |
| 15:47.18 | Gido-E | dlynes my experience with fax and voip is good. Although i always point to the fact, it sould not work that good :-) |
| 15:47.33 | tty | hi. i am trying to set up asterisk on my mac. so that i can test sipdroid from my android phone.. i cant get asterisk to work |
| 15:48.00 | dlynes | ~mac |
| 15:48.00 | infobot | [mac] ... MAC?!???! *chomp* |
| 15:48.03 | dlynes | ~bsd |
| 15:48.04 | infobot | BSD is a UNIX operating system. An asterisk port is currently availible if you feel you must, or a way to set your pc back 30 years, progress is overrated |
| 15:48.27 | beers218 | Gido-E: what version of asterisk do you run? |
| 15:48.32 | Gido-E | 1.4 |
| 15:48.36 | tty | :P. yes mac. it ijust for testbed. |
| 15:48.39 | coppice | ManxPower-work: there are many servers which handle hundreds of thousands of FAXes per day (PSTN, not SIP). It can work, though any setup needs thorough testing |
| 15:48.45 | Gido-E | 1.6 is to new for me :-) |
| 15:48.57 | ManxPower-work | tty: no, it will be a learning experience for you since nobody here will be able to help you. |
| 15:48.58 | beers218 | you must have nvfax working then |
| 15:49.04 | Gido-E | agx-fax |
| 15:49.23 | [TK]D-Fender | tty: Could you be a little more vague please? |
| 15:49.36 | beers218 | well that works too... 1.6 doesnt support agx or nv |
| 15:49.54 | Gido-E | is there a good fax implemention for 1.6? |
| 15:49.57 | beers218 | that or i couldnt find anything to make it work |
| 15:51.13 | beers218 | there is fax for asterisk, but im having issues with it ATM |
| 15:51.21 | dlynes | tty, it works on freebsd with ports tree, but even then doesn't work as well as it does on linux....mac has a few dependencies that are different from standard bsd |
| 15:51.26 | ManxPower-work | Gido-E: the one that comes with 1.6 isn't good enough for you? |
| 15:51.32 | *** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler) |
| 15:52.05 | tty | [TK]D-Fender: i want to use sipdroid (application for android OS) to connect to a SIP server and be able to open a rtsp video. so i am trying to use sipdroid with VoIP first. I tried with bpxes.org/com but i cannot connect! so i assumed why not set up my own SIP Server (Asterisk) and try it out. :) what do you think? |
| 15:52.12 | Gido-E | ManxPower-work actually: yes... |
| 15:52.28 | ManxPower-work | Gido-E: then Digium has a commercial fax solution for you |
| 15:52.53 | [TK]D-Fender | tty: I think you have not described what your PROBLEM with using Asterisk is. |
| 15:52.54 | tty | [TK]D-Fender: i am having a macbook here (unfortunatelly no linux pc around) |
| 15:53.02 | Gido-E | ManxPower-work Thnx for the tip. |
| 15:53.12 | tty | [TK]D-Fender: my problem is i am tryin to set up Asterisk on my macbook |
| 15:53.34 | dlynes | tty, that's not a problem if you try it, it installs and it works...so what exactly is the problem? |
| 15:53.41 | Gido-E | mac has parralels ? |
| 15:53.42 | ManxPower-work | tty: no. That is what you are trying to DO. Your SPECIFIC problem is unknown since you have not told us WHAT IS NOT WORKING? |
| 15:53.53 | [TK]D-Fender | tty: I'd hate to be your mechanic. Your entrire car would be taken apart and rebuilt for NOTHING because when you said "my car is broke", is discovered to really mean "the little plastic tab that holds my mug in the cupholder is broken" |
| 15:54.00 | dlynes | Gido-E, you mean plesk parallels? yes, it does |
| 15:54.01 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 15:54.18 | tty | i am not sure how to use a SIP server.. (asterisk) |
| 15:54.27 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 15:54.37 | Gido-E | install asterisk under a linux env. under parallels than? |
| 15:54.41 | [TK]D-Fender | tty: That's like saying.... "How do I drive". Gor ead the book. |
| 15:54.45 | [TK]D-Fender | ~book |
| 15:54.46 | 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 |
| 15:54.47 | [TK]D-Fender | ^^^^^^^^^^^^^ |
| 15:54.51 | dlynes | ~thebook |
| 15:54.52 | infobot | hmm... thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org |
| 15:55.10 | tty | that's i ll check it out :) |
| 15:55.15 | ManxPower-work | tty: Asterisk is not a "sip server". Asterisk is a PBX Toolkit that lets you build a multiprotocol PBX. |
| 15:55.58 | *** join/#asterisk hfb (n=hfb@96.247.53.139) |
| 15:56.31 | tty | ManxPower-work: hmmmmm |
| 15:57.31 | tty | ManxPower-work: so, there is no SIP capabilities in asterisk? |
| 15:57.40 | *** join/#asterisk etnos (n=chatzill@host-208-88-126-198.biznesshosting.net) |
| 15:57.43 | etnos | G'morning everyone |
| 15:58.06 | minotaur01 | morning |
| 15:58.19 | [TK]D-Fender | tty: There are. Now go read the BOOK |
| 15:58.26 | tty | ManxPower-work: ;) |
| 16:03.18 | ManxPower-work | tty: "MULTI protocol" |
| 16:03.57 | *** join/#asterisk Rabenklaue (n=Rabe@p54A7F46D.dip.t-dialin.net) |
| 16:04.27 | tty | ManxPower-work: so i will be able to send a rtsp video through asterisk, right? |
| 16:05.04 | Rabenklaue | I do not use my asterisk installation that lot, but I think after the update from 1.6.0.XX to 1.6.1.XX I get following error: |
| 16:05.07 | Rabenklaue | [Jan 19 16:58:53] WARNING[3329]: app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) |
| 16:05.16 | Katty | considers lunch possibilities. |
| 16:05.18 | Rabenklaue | While trying to call a remote number. |
| 16:05.21 | ManxPower-work | tty: less talk, more reading of The Book |
| 16:05.30 | Rabenklaue | With 1.6.2 it's the same |
| 16:05.40 | tty | ManxPower-work: got it :P |
| 16:05.52 | Rabenklaue | I did not change anything in my extensions.conf |
| 16:06.17 | ManxPower-work | Rabenklaue: your DAHDI install is broken. |
| 16:06.19 | Rabenklaue | "DAHDI Version: 2.2.0.2 Echo Canceller: KB1" with zaphfc |
| 16:06.39 | bmoraca_work | Rabenklaue, did you recompile DAHDI after you upgraded asterisk? |
| 16:06.55 | Rabenklaue | http://pastebin.ca/1757584 |
| 16:07.10 | Rabenklaue | bmoraca_work: No I did not recompile dahdi |
| 16:07.16 | ManxPower-work | Rabenklaue: does "lsmod" show dahdi loaded? |
| 16:07.42 | casix | I have a meetme with 2 users, one is marked, the meetme is started with x option (close the conference when last marked user exits) I wan to continue with the other user (the one that is not marked) but the channel hangsup. Is possible to continue with the channel? |
| 16:07.53 | Rabenklaue | ManxPower-work: http://pastebin.ca/1757585 |
| 16:08.23 | ManxPower-work | Rabenklaue: your problem is in /etc/asterisk/chan_dahdi.conf |
| 16:08.30 | Rabenklaue | ManxPower-work: It worked before |
| 16:08.57 | ManxPower-work | Rabenklaue: then roll back to what you were using. |
| 16:09.18 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 16:09.37 | Rabenklaue | chan_dadhi.conf: http://pastebin.ca/1757587 |
| 16:10.08 | ManxPower-work | Rabenklaue: pastebin the output of "dahdi show channels" |
| 16:10.27 | *** join/#asterisk Alagar (n=Administ@122.164.34.38) |
| 16:10.52 | ManxPower-work | rememebr you set the OPTIONS THEN you specify the channel => none of your options canpark=yes, localprefix= etc are being used because they come after the channel => line. |
| 16:10.55 | Rabenklaue | ManxPower-work: http://pastebin.ca/1757594 |
| 16:11.13 | ManxPower-work | Rabenklaue: now pastebin the cli output of a failed call |
| 16:12.24 | *** join/#asterisk p3nguin (i=gpz5GvdF@9xf6gbZ.a2infotech.com) |
| 16:13.16 | Rabenklaue | ManxPower-work: http://pastebin.ca/1757595 |
| 16:13.23 | [TK]D-Fender | Rabenklaue: Show use the COMPLETE failed call, and your configs |
| 16:13.33 | *** join/#asterisk Heretic (n=fallen@dsl-246-71-178.telkomadsl.co.za) |
| 16:13.52 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.229.58) |
| 16:13.57 | TommyBotten | Why does dahdi load the us toneset, even if I don't have it configured? |
| 16:14.00 | ManxPower-work | Rabenklaue: you are trying to dial g1, but you have no group=1 set for those channels (the group line is after the channel => line so it's not used for that channels(s) |
| 16:17.39 | *** join/#asterisk timahvo1 (n=rogue@41.191.224.178) |
| 16:17.41 | Rabenklaue | The updated chan_dahdi.conf: http://pastebin.ca/1757605 |
| 16:18.23 | ManxPower-work | Rabenklaue: you may not wabt faxdetect=both, but other than that it looks good. You may have to unload/load chan_dahdi.so for the change to take effect. |
| 16:18.47 | Rabenklaue | http://pastebin.ca/1757607 |
| 16:19.27 | Rabenklaue | ManxPower-work: I want to use hylafax later to send and receive fax. Then I would need this entry, or? |
| 16:19.43 | ManxPower-work | [Jan 19 17:17:53] WARNING[4810]: app_dial.c:1547 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) The telco is rejecting the call now. Before Asterisk wasn't even sending the call to the telcol |
| 16:20.29 | ManxPower-work | Rabenklaue: faxdetect will make asterisk hijack all calls with a fax tone and try to them them to the exten => fax extension. Is that what you want, even for faxes you are trying to SEND? |
| 16:21.32 | ManxPower-work | Rabenklaue: do not put a 0 at the beginning of the calleridnum |
| 16:22.59 | TommyBotten | What is the difference between groundstart, loopstart og kewl/koolstart? |
| 16:25.53 | ChannelZ | http://lmgtfy.com/?q=ground+start+vs+loop+start |
| 16:26.37 | *** join/#asterisk RobH (n=robh@rob.tech.wikimedia.org) |
| 16:26.39 | TommyBotten | I figured... I thought it was written in one word |
| 16:27.25 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 16:27.33 | [TK]D-Fender | Rabenklaue: Some telcos respond with 34 to say the dest # is busy |
| 16:28.40 | Rabenklaue | [TK]D-Fender: It does not matter what number I try to call, everytime the 34 error is displayed |
| 16:28.58 | *** join/#asterisk ashd (n=ashleydr@94-194-208-216.zone8.bethere.co.uk) |
| 16:29.16 | [TK]D-Fender | Rabenklaue: Check to make sure your channels are free and call your telco to confirm the message on their end |
| 16:30.34 | Rabenklaue | http://pastebin.ca/1757620 |
| 16:30.56 | Rabenklaue | Does this In Service mean, that the channels are in use right now? |
| 16:31.41 | ChannelZ | No |
| 16:31.52 | *** join/#asterisk mlr78 (n=mratliff@74.5.197.138) |
| 16:32.05 | ChannelZ | Just that they are functional |
| 16:32.45 | mlr78 | what tool do you recommend for editing gsm files? |
| 16:33.31 | mlr78 | I've tried audacity, but gsm isn't supported. I'd like to be able to create and/or edit gsm files to be used with asterisk |
| 16:34.00 | Naikrovek | why do you want to edit gsm directly? convert to WAV and (gulp) reencode |
| 16:34.01 | ChannelZ | It's easier to just work in WAV and then convert to gsm later |
| 16:34.27 | Naikrovek | avoids GSM entirely |
| 16:35.10 | Naikrovek | that reencode is going to sound AWFUL though |
| 16:35.23 | [TK]D-Fender | mlr78: http://manual.audacityteam.org/index.php?title=File_Export_Dialog |
| 16:35.34 | [TK]D-Fender | mlr78: Audacity DOES support GSM 6.10 |
| 16:35.51 | [TK]D-Fender | mlr78: And do not WORK in GSM, Export your final product to it |
| 16:36.37 | ChannelZ | Naikrovek: why would it sound any more awful than gsm on it's own? |
| 16:37.58 | [TK]D-Fender | Never START with GSM |
| 16:37.58 | tzafrir_laptop | It's also better to edit the original file (if available) and convert to gsm once you're done |
| 16:38.07 | [TK]D-Fender | ^^^ |
| 16:38.33 | tzafrir_laptop | mlr78, you can easily convert e.g. wav files to gsm using sox |
| 16:41.43 | *** join/#asterisk titter (n=titter@c-76-101-240-142.hsd1.fl.comcast.net) |
| 16:42.04 | *** join/#asterisk seather (n=seather@196.22.137.18) [NETSPLIT VICTIM] |
| 16:42.04 | *** join/#asterisk Dibbler (n=Dibbler@87.194.103.72) [NETSPLIT VICTIM] |
| 16:42.06 | *** join/#asterisk dennis00 (n=user@unaffiliated/dennis00) |
| 16:42.21 | *** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) [NETSPLIT VICTIM] |
| 16:43.31 | minotaur01 | has anyone come across a problem where asterisk dosn't load the config files properly on a restart... but dose load them properly on a reload? |
| 16:43.41 | dennis00 | How do I check if I can use callerid with voipbuster without any credit left? |
| 16:46.16 | *** join/#asterisk luckyaba (n=lucky@ip72-194-218-169.sb.sd.cox.net) |
| 16:46.19 | Chainsaw | Qwell: Quickest way I can do that for you is by giving you a full log of what the ebuild does. That okay with you? (make plus signs) |
| 16:46.27 | *** join/#asterisk GabrielPiassetta (n=gabriel@201.47.45.199.dynamic.adsl.gvt.net.br) |
| 16:47.06 | Qwell | Chainsaw: that works. with and without the patch would be great |
| 16:47.16 | Chainsaw | Qwell: *nod* Will do. |
| 16:48.09 | *** join/#asterisk sebbl (n=Momofu@HSI-KBW-078-043-193-153.hsi4.kabel-badenwuerttemberg.de) |
| 16:50.20 | *** join/#asterisk shazaum (n=shazaum@201.47.74.147) |
| 16:50.21 | *** join/#asterisk benngard (n=benngard@90-230-92-67-no148.tbcn.telia.com) |
| 16:50.35 | GabrielPiassetta | there some place where I can get the source rpm package of asterisk? |
| 16:50.59 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:51.02 | Qwell | GabrielPiassetta: The one provided by Digium for AsteriskNOW? |
| 16:51.22 | *** part/#asterisk shazaum (n=shazaum@201.47.74.147) |
| 16:51.22 | Qwell | http://packages.asterisk.org/centos/5/current/SRPMS/ |
| 16:51.31 | GabrielPiassetta | thank you |
| 16:53.18 | *** join/#asterisk lynxsys (n=thelynx@82-71-19-61.dsl.in-addr.zen.co.uk) |
| 16:53.31 | lynxsys | has anybody successfully setup QOS for voice over a single broadband connection in the UK? |
| 16:53.32 | *** join/#asterisk high-freq (n=hfreq@adsl-99-41-142-239.dsl.ksc2mo.sbcglobal.net) |
| 16:54.09 | Chainsaw | Qwell: Attached. Most visible in the install stage. |
| 16:54.23 | *** join/#asterisk CunningPike (n=CunningP@204.239.8.157) |
| 16:54.41 | ManxPower-work | lynxsys: that is impossible without carrier support./ |
| 16:56.29 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 16:57.26 | lynxsys | ManxPower-work: are you in the UK? |
| 16:57.47 | ManxPower-work | lynxsys: no, but I know QoS and if you want real QoS it must be configured on both ends like the link |
| 16:58.16 | lynxsys | ok wondered if you were familiar with 21CN |
| 16:58.30 | Chainsaw | lynxsys: That's BT-speak for "stack of cheap Huawei routers". |
| 16:58.48 | lynxsys | lol |
| 16:58.48 | titter | anyway to list all conferences at once? |
| 16:59.03 | titter | i can do meetme list exten |
| 16:59.07 | lynxsys | Chainsaw: will it support QoS? |
| 16:59.20 | *** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com) |
| 16:59.25 | Chainsaw | lynxsys: The underlying kit will. |
| 16:59.34 | Chainsaw | lynxsys: The ways of BT management and call centre agents can not be predicted. |
| 17:00.17 | *** join/#asterisk Badrobot- (n=badrobot@cpe-76-173-229-89.socal.res.rr.com) |
| 17:00.21 | lynxsys | Chainsaw: true, but will it give my "carrier" the ability to use QoS over my broadband line? |
| 17:00.47 | Chainsaw | lynxsys: Your "carrier" will be buying a product called WDC from BT. |
| 17:01.01 | Chainsaw | lynxsys: So again, that is up to BT. If you do QoS right, every component in the chain supports it. |
| 17:01.16 | Poincare | anyone in here who has any experience with 'easyphone GSM over IP' stuff? |
| 17:01.25 | *** join/#asterisk Cain` (n=Geek@unaffiliated/cain) |
| 17:01.55 | Amorsen | What is WDC? |
| 17:02.14 | Qwell | Chainsaw: I'm only having to change 1 line (where it calls $(SUBMAKE)) to get rid of the errors |
| 17:03.13 | Chainsaw | Qwell: Then by all means do less changes. If I still get errors, I'm happy to open a new bug for 1.6.2.2 |
| 17:03.14 | Qwell | That should be the only change required, since that's the only instance of make being called that differs from $(MAKE) |
| 17:03.22 | Qwell | Could you try it with just that one change? |
| 17:03.28 | Qwell | +@+DESTDIR="$(DESTDIR)" ASTSBINDIR="$(ASTSBINDIR)" $(SUBMAKE) -C $(@:-install=) install |
| 17:03.32 | Qwell | that line, specifically |
| 17:03.47 | Amorsen | I've rarely seen a less useful google search than that for WDC |
| 17:03.57 | lynxsys | Chainsaw: what is WDC? |
| 17:04.10 | Qwell | all of the other ones do actually call $(MAKE) (except the one in menuselect/Makefile on the./configure line, which I think is incorrect to change) |
| 17:05.40 | Chainsaw | Amorsen, lynxsys: WBC, apologies. |
| 17:05.43 | *** join/#asterisk Kobaz (n=kobaz@its.kobaz.net) |
| 17:06.13 | lynxsys | Chainsaw: lol ok what is WBC? |
| 17:06.29 | Kobaz | okay soooo |
| 17:06.39 | Kobaz | i have a meetme... and it's taking calls over t1/dahdi |
| 17:06.39 | Chainsaw | lynxsys: "Wholesale Broadband Connect" : http://www.btwholesale.com/pages/static/Products/Broadband/Wholesale_Broadband_Connect_WBC.html |
| 17:07.07 | Kobaz | if someone hasn't been talking in a a little bit... and starts talking... there's always a 'buzz' during the first few ms of their speaking |
| 17:07.36 | Chainsaw | Qwell: Just the submake, k. Trying. |
| 17:07.37 | Amorsen | Thanks |
| 17:08.18 | Qwell | Chainsaw: there might be an issue in uninstall too, for the same reason |
| 17:08.27 | Qwell | yes, there is |
| 17:08.36 | lynxsys | thanks |
| 17:08.47 | Kobaz | and i'm not using talking optimization 'o' |
| 17:08.58 | Qwell | <PROTECTED> |
| 17:09.03 | Qwell | Chainsaw: that line needs changed too |
| 17:09.11 | [TK]D-Fender | [12:07]<Kobaz>if someone hasn't been talking in a a little bit... and starts talking... there's always a 'buzz' during the first few ms of their speaking <- the Steve Jobs Effect |
| 17:09.19 | Kobaz | heh |
| 17:09.20 | Kobaz | what? |
| 17:09.25 | Kobaz | it's a buzz in the audio |
| 17:09.26 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 17:09.27 | Kobaz | like static |
| 17:09.37 | [TK]D-Fender | Kobaz: Could be jitter before the buffers hit it |
| 17:09.43 | Amorsen | I wonder how difficult it is to get a WBC deal... But then I guess our foreign expansion has been put on hold for a while. |
| 17:10.01 | Kobaz | it's all tdm |
| 17:10.09 | Kobaz | there's a jitter buffer in use stuff? |
| 17:10.14 | Kobaz | stuff/still |
| 17:11.23 | Chainsaw | Qwell: Compile in progress, will have results in ~4 minutes. |
| 17:11.51 | Chainsaw | lynxsys: WBC is already being sold. Any ISP offering BT-based ADSL faster than 8mbit is using it. |
| 17:12.01 | *** join/#asterisk xpot-mobile (n=xpot@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net) |
| 17:12.08 | Chainsaw | lynxsys: (Because that's ADSL2/ADSL2+ and the non-21CN BT equipment can not handle that) |
| 17:12.56 | Kobaz | [TK]D-Fender: weird problem, eh? |
| 17:14.00 | [TK]D-Fender | Kobaz: TDM on both sides? |
| 17:14.29 | Kobaz | yes |
| 17:14.53 | [TK]D-Fender | Kobaz: Perhaps PCI latency, etc... |
| 17:14.58 | Kobaz | mm |
| 17:15.00 | Kobaz | how would i fix that? |
| 17:15.01 | Kobaz | heh |
| 17:15.03 | [TK]D-Fender | Kobaz: What card? |
| 17:15.07 | Kobaz | sangoma a102 |
| 17:15.18 | [TK]D-Fender | Kobaz: I'd call them up on it. |
| 17:15.31 | Kobaz | don't think it's meetme() ? |
| 17:15.56 | Kobaz | i can try tdm to tdm calls, just bridged, no meetme and see if the same issue persists |
| 17:16.39 | bmoraca_work | Kobaz, make sure none of your endpoints have VAD enabled...what you describe is EXACTLY what would happen with that enabled (low quality audio in the first few MS of speech) |
| 17:17.05 | Kobaz | hmm |
| 17:17.08 | bmoraca_work | well, low quality of missing audio |
| 17:17.10 | Kobaz | is that a feature on avaya systems? |
| 17:17.13 | bmoraca_work | or* |
| 17:17.17 | bmoraca_work | Kobaz, it could me |
| 17:17.19 | [TK]D-Fender | Kobaz: Oh you mean NOW you're going to actually pin the scope of the problem down? :p |
| 17:17.24 | Kobaz | [TK]D-Fender: haha |
| 17:17.28 | Kobaz | [TK]D-Fender: yeah :P |
| 17:18.02 | Kobaz | well i thought someone might have said something what bmoraca said... something like "oh yeah, i've seen that before" |
| 17:18.06 | Chainsaw | Qwell: Confirmed, I will post your minimal patch instead. I carried this all over from 1.6.1 which needed more love before the warnings went away. |
| 17:18.36 | Qwell | k. do you understand now why only those 2 lines are required? |
| 17:19.20 | Chainsaw | Qwell: As I said, most of this was developed on 1.6.0 & 1.6.1; 1.6.0 is particular spewed at least 20 warnings. |
| 17:19.23 | Qwell | When the $(MAKE) variable is used (anywhere), it can do things correctly |
| 17:19.32 | Qwell | in theory, anyways :D |
| 17:20.28 | Chainsaw | Yeah, I'm one of those "I'll believe it when I see it" people. And I saw. So you win :) |
| 17:20.29 | Qwell | find . -name 'Makefile' -exec grep -E "SUBMAKE|SILENTMAKE" {} \; |
| 17:20.31 | Chainsaw | Posted, please apply. |
| 17:20.38 | Qwell | That should tell you which lines are bad, in any given branch |
| 17:20.57 | Qwell | (if there's no +, it's probably wrong) |
| 17:25.35 | *** join/#asterisk citywok (n=chatzill@vpn.csgopenline.com) |
| 17:25.48 | citywok | what is channel.c:1040 __ast_queue_frame: Exceptionally long voice queue length queuing to Local/1155@inside-leg-1f34;1 |
| 17:25.50 | citywok | ? |
| 17:26.32 | *** join/#asterisk sbrath (n=sbrath@unaffiliated/sbrath) |
| 17:27.07 | leifmadsen | citywok: what version of Asterisk? |
| 17:27.19 | citywok | 1.6.1.12 |
| 17:27.27 | leifmadsen | citywok: if you search for that in the issue tracker, you will find an issue for that -- I think it was fixed in 1.6.1.13 |
| 17:27.56 | *** join/#asterisk paulc (n=paulc@unaffiliated/paulc) |
| 17:28.10 | sbrath | If I have my usernames for phones as a number, dialing 7777 takes me right to the mailbox. If I change the usernames to be different, dialing 7777 takes me to a "Welcome to the voicemail system enter your mailbox".. How do I change that? |
| 17:28.19 | citywok | ooooooh |
| 17:28.26 | citywok | "This phenomenon also occurs if you have a queue with 'ringall' strategy and the members are all on 'Local/' channels. |
| 17:28.28 | citywok | If two members answer the call at the same time, this deadlock occurs and no further calls can be processe" |
| 17:28.34 | leifmadsen | M15609 |
| 17:28.43 | MuffinMan | [closed] [Asterisk] Core/Channels 0015609: [patch] WARNING[23025]: channel.c:952 __ast_queue_frame: Exceptionally long voice queue length queuing to Local reported by aragon https://issues.asterisk.org/view.php?id=15609 |
| 17:28.44 | citywok | that's plausible |
| 17:28.47 | leifmadsen | citywok: see above |
| 17:29.00 | citywok | yea, i was reading that bug report, that comment i posted came from there |
| 17:29.05 | leifmadsen | gotcha |
| 17:29.32 | citywok | i was running SVN until last week, before going back to 1.6.1.12 so thats probably why it started happening |
| 17:29.33 | leifmadsen | sbrath: you need to pass the username/mailbox number to the VoicemailMain() application |
| 17:29.37 | [TK]D-Fender | sbrath: Go look at your dialplan |
| 17:29.41 | leifmadsen | citywok: ah gotcha |
| 17:29.48 | leifmadsen | citywok: well there ya go :) |
| 17:29.55 | [TK]D-Fender | sbrath: It goes where you told it to |
| 17:29.56 | leifmadsen | ok, lunch time! |
| 17:29.58 | *** join/#asterisk fofware (n=chatzill@190.7.25.160) |
| 17:30.05 | citywok | the svn version was having issues for me so i decided to go do it the right way since my last changes that were show stopping were committed to .12 |
| 17:30.28 | *** join/#asterisk ashd (n=ashleyd@94-194-208-216.zone8.bethere.co.uk) |
| 17:30.34 | leifmadsen | citywok: well, don't be afraid to file issues, especially if they are regressions |
| 17:31.16 | citywok | alright, thanks! |
| 17:37.51 | *** join/#asterisk ttl- (n=patrick@d5153A420.access.telenet.be) |
| 17:43.11 | *** join/#asterisk RobH (n=robh@cpe-173-169-30-118.tampabay.res.rr.com) |
| 17:43.27 | *** join/#asterisk ashd (n=ashleyd@94-194-208-216.zone8.bethere.co.uk) |
| 17:44.48 | *** join/#asterisk Strogg (n=jean@unaffiliated/strogg) |
| 17:44.51 | *** join/#asterisk SmartPlan (n=office@92.86.198.21) |
| 17:45.27 | SmartPlan | hello could any one please help me with installing asterisk? |
| 17:45.34 | SmartPlan | it is my first time... |
| 17:45.46 | SmartPlan | my single problem... |
| 17:45.48 | SmartPlan | is one error |
| 17:45.51 | Qwell | SmartPlan: tar xvf asterisk-*.tar.gz; cd asterisk; ./configure; make; make install |
| 17:46.07 | SmartPlan | i get this error |
| 17:46.08 | SmartPlan | Failed to connect to the Asterisk manager through port: 5038 |
| 17:46.19 | Kobaz | SmartPlan: i take paypal, checks, money orders, cookies, muffins, happy endings (from females) |
| 17:46.29 | SmartPlan | and Reload failed because retrieve_conf encountered an error: 1 |
| 17:46.35 | Qwell | SmartPlan: #freepbx |
| 17:46.37 | voipmonk | looks like freepbx |
| 17:46.38 | path | maybe the manager is not listening :) |
| 17:46.46 | sbrath | [TK]D-Fender: I was passing the CID as the mailbox number to the VoiceMailApp I'll do some debugging to see if that isn't being set as I thought it was. |
| 17:46.53 | path | looks like asterisk-gui |
| 17:46.56 | Qwell | path: managers rarely listen |
| 17:47.03 | SmartPlan | ok thanks |
| 17:47.05 | path | Qwell: indeed |
| 17:47.13 | Chainsaw | Qwell: Thank you :) |
| 17:47.25 | Qwell | Chainsaw: That'll be $99.95 |
| 17:47.30 | Qwell | (not really) |
| 17:47.40 | sbrath | SmartPlan: What did you install? AsteriskNOW, or are you installing from source.? |
| 17:47.44 | Chainsaw | *G* |
| 17:48.09 | Qwell | Chainsaw: I just wanted to make sure I was fixing the right thing |
| 17:48.17 | Chainsaw | Qwell: It makes sense, yes. |
| 17:48.30 | SmartPlan | <sbrath>: i installed AsteriskNow, and those are the only errors i get |
| 17:48.44 | Qwell | What I was initially doing was`./configure; make -j8 install`, and that was...erm...horribly broken. |
| 17:49.12 | Kobaz | Qwell: parallell builds work fine for me |
| 17:50.16 | Kobaz | Qwell: what asterisk version? |
| 17:50.47 | Chainsaw | Qwell: This was about 1.6.2.1 |
| 17:50.53 | Chainsaw | Kobaz: They work, but it whines a little during install. |
| 17:51.01 | Qwell | whined* |
| 17:51.08 | Kobaz | oh |
| 17:51.45 | Chainsaw | Kobaz: Unless you do install with -j of course, but that's not standard practise in Gentoo Linux. |
| 17:51.53 | Chainsaw | without -j, even. |
| 17:52.20 | *** join/#asterisk hfb (n=hfb@96.247.53.139) |
| 17:54.44 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 17:58.12 | Kobaz | mm |
| 17:58.42 | Strogg | 'lo 'lo. I'm trying to set my caller Id so it shows up properly. Right now when I call someone/bm 10 |
| 17:58.48 | bmoraca_work | lol...almost 21 hours of talk time on one of my DS0s |
| 17:59.00 | Strogg | GAhh. mispost. was trying to change windows midway through question. Heh. |
| 18:00.36 | Strogg | 'lo 'lo. I'm trying to set my caller Id so it shows up properly. Right now when I call someone, their caller id says unknown number. I have the Set(CALLERID(all)=First Last <my number>) in my dialplan, and I have the app_setcallerid.so and func_callerid.so modules loaded. What am I missing? |
| 18:01.22 | ManxPower-work | Strogg: Have you confirmed your carrier allows you to set the callerid. Also, only callerid number is accepted by the PSTN, not name |
| 18:01.48 | ManxPower-work | also what EXACTLY are you setting the CLID number to? |
| 18:02.18 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 18:07.11 | *** join/#asterisk binbash_ (n=peter@ip4da53781.direct-adsl.nl) |
| 18:07.18 | Akiraa | Is it normal for ATA devices (FXO) to have no credentials for the PBX to gain PSTN access? Otherwise, is it normal for an ATA device with FXO and FXS ports to have the same credentials for the FXS user account as the IPPBX credential for PSTN outbound trunks? |
| 18:07.26 | Strogg | ManxPower-work: I have a DID (unlimitel) that gives us an AIX connection and a phone number.. they have a configuration generator which specifies the callerid line.. |
| 18:08.16 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 18:08.25 | Strogg | ManxPower-work: for the caller id line.. I have.. exten => _NXXNXXXXXX,n,Set(CALLERID(name)=MyCompany) |
| 18:09.03 | *** join/#asterisk Tech_Travis (n=tech_tra@208.179.137.131) |
| 18:09.05 | Strogg | ManxPower-work: I have also tried exten => _NXXNXXXXXX,n,Set(CALLERID(all)=My Company <2345678900>) |
| 18:09.27 | Strogg | it's the 2nd rule in my dialplan for dialing out.. hence the 'n' |
| 18:09.56 | bmoraca_work | Strogg, a pastebin with your entire outbound dial context would be more helpful. |
| 18:10.33 | bmoraca_work | Strogg, additionally, callerid name is looked up by the CALLED party and cannot be influenced at call-time by the calling party (except in certain "on-net" situations) |
| 18:10.37 | ManxPower-work | Strogg: I've never seen a carrier example asterisk config that actually was correct. |
| 18:11.00 | ManxPower-work | BTW, does anyone know of a site to look up the company that has a specific toll free number in the usa? |
| 18:11.08 | Strogg | well I wouldn't be surprised if it has errors.. but I'm guessing they support it, since they put the feature in there. |
| 18:12.31 | Strogg | bmoraca_work: ManxPower-work: http://pastey.net/131685-uowv |
| 18:12.39 | Strogg | here's the local dialing section |
| 18:13.13 | ManxPower-work | Strogg: contact your carrier. They are the ones not alllowing the outgoing callerid. your callerid setting is correct. |
| 18:13.20 | Katty | had fazoli's for lunch (= |
| 18:13.27 | Strogg | Hrmm Ok. |
| 18:13.30 | Strogg | ManxPower-work: thanks.. I'll do that |
| 18:15.40 | Naikrovek | what are good embedded platforms for asterisk people have used? my decade old laptop went kaput I was using for asterisk went kaput over the weekend |
| 18:15.51 | Naikrovek | suppose i could see if my router is supported for openwrt |
| 18:17.49 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 18:18.38 | *** join/#asterisk jaskew (n=jdaskew@netblock-66-159-217-102.dslextreme.com) |
| 18:22.40 | *** join/#asterisk retentiveboy (n=pdugas@adsl-221-170-128.pns.bellsouth.net) |
| 18:23.37 | bmoraca_work | Naikrovek, i've been looking at something like this: http://www.newegg.com/Product/Product.aspx?Item=N82E16816101262 ....it's not embedded, but it's definitely small enough and could probably be run fanless with an SSD or CF drive |
| 18:25.26 | *** join/#asterisk hluesea (n=hulusika@88.247.127.66) |
| 18:25.28 | Naikrovek | think i'm going to piece together some kind of mini-itx system and use it for asterisk |
| 18:26.03 | Naikrovek | suppose i could boost some disused hardware from work |
| 18:26.05 | Naikrovek | in a pinch |
| 18:26.32 | *** join/#asterisk n3hxs (n=HAMming@65-121-75-131.dia.static.qwest.net) |
| 18:26.32 | bmoraca_work | i looked at doing that a while back...and honestly, that supermicro case is both slimmer and more powerful than any mini-itx i was able to readily find parts for |
| 18:26.43 | Naikrovek | okay good to know |
| 18:26.52 | bmoraca_work | you can rotate the rack ears 90-degrees and use them to screw it to a wall |
| 18:26.55 | Naikrovek | i just don't want to buy an entire laptop just for this |
| 18:26.55 | *** join/#asterisk markwaters (n=mark@weloveit.info) |
| 18:27.08 | *** join/#asterisk bsaxon (n=bsaxon@12.68.234.174) |
| 18:27.57 | citywok | is it possible to sipaddheader from the AMI when originating an outbound call? |
| 18:29.10 | bmoraca_work | not directly, now...but you can specify variables in AMI Originations and use SIPAddHeader in your dialplan context |
| 18:29.24 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.243) |
| 18:29.49 | citywok | i'm trying to originate a call from an extension, and have it auto-answer using the header for that |
| 18:29.54 | citywok | without using local channels |
| 18:30.20 | citywok | local channels mess with chanspy()'s ability to announce the name of the person being spied on |
| 18:30.35 | bmoraca_work | good luck. |
| 18:30.45 | citywok | lol, that doesnt sound promising |
| 18:31.10 | bmoraca_work | i don't know of a way to do it, but i don't believe it's possible. |
| 18:31.11 | Corydon76-dig | You're trying to do what the Local channel was designed to do... without using the Local channel. Good luck! |
| 18:31.32 | citywok | yea, local channel is what i use now, but i dislike that it will not announce the name of the person i'm listening to in chanspy |
| 18:32.19 | Corydon76-dig | That would be because it's on the other side of the local channel |
| 18:32.56 | Corydon76-dig | A local channel is actually 2 channels, with automatic passing of frames between them |
| 18:33.19 | citywok | yea, a local channel call ends up using 3 channels, 1 to the call from, the local channel, and 1 to the call to |
| 18:34.15 | Corydon76-dig | No, actually 4. |
| 18:34.30 | citywok | oh, whooops, you are right |
| 18:34.42 | citywok | a channel for the local call out, and a channel for the local call in |
| 18:35.19 | citywok | so is there a way to make chanspy listen to the inside leg instead of hte outside leg? would it then even be able to announce that persons voicemail name? |
| 18:36.11 | Corydon76-dig | Technically, yes |
| 18:36.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 18:36.27 | citywok | using the chanspy filter? chanspy(local/????) |
| 18:36.31 | ruben23 | hi anyone use snom IP phones for asterisk..? |
| 18:37.05 | Amorsen | Yes |
| 18:37.59 | *** join/#asterisk slinksh0t (n=slinksh0@74.115.208.59) |
| 18:43.56 | mlr78 | when I dial into an extension and reach voicemail, then hang up, asterisk doesn't recognize that I've hung up and keeps recording. Any way to force a timeout? |
| 18:44.08 | jaskew | I'm cleaning up a page on voip-info.org and want to make sure I don;t do anything stupid... Has Free World Dialup passed into fond memory (i.e. are they dead)? |
| 18:44.35 | citywok | mlr78: what type of channels? |
| 18:46.04 | mlr78 | t1 |
| 18:47.34 | citywok | make sure all your signaling is turned off |
| 18:47.51 | citywok | turned off? errr, set properly* |
| 18:48.00 | citywok | sounds like your channels aren't being closed properly when they are hung up |
| 18:49.07 | *** join/#asterisk TimeRider (n=steve@78.32.26.1) |
| 18:50.03 | *** join/#asterisk ticoit (n=ticoit@201.191.187.208) |
| 18:51.20 | Amorsen | Was that all you wanted to know ruben23? |
| 18:52.57 | ruben23 | Amorsen:have anyone used snom IP phones for asterisk |
| 18:53.28 | ManxPower-work | ruben23: plenty of SNOM info on voip-info.org. Some of it may be wrong or out of date. |
| 18:55.44 | ManxPower-work | hugs his Polycom phone |
| 18:56.39 | citywok | hugs his aastra phone |
| 18:56.58 | citywok | chucks his aastra cordless handset out the window for being a piece of shit afterthought |
| 18:56.59 | p3nguin | kisses his Cisco phone |
| 18:57.44 | ruben23 | cisco is mostly used right..? |
| 18:57.58 | Qwell | Cisco is abhorred. |
| 18:58.07 | jaskew | apparently p3nguin's has... |
| 18:58.11 | p3nguin | I don't know about mostly, but some people use Cisco. |
| 18:58.20 | citywok | yea i dislike the cisco phones i tried out |
| 18:58.29 | p3nguin | What models? |
| 18:58.31 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 18:58.39 | citywok | 7960, 7970GE |
| 18:58.50 | p3nguin | How could you not like a 7960? |
| 18:58.51 | citywok | the color 7970 looked pretty, but i wouldnt want to have to configure it |
| 18:59.07 | citywok | the 7960 was easier to configure, but really disliked any form of NAT |
| 18:59.38 | citywok | the speakerphone was okay on it, the aastra had hte best speaker out of all the phones |
| 18:59.51 | jaskew | I have a couple Aastra 480i phones. people say the sound quality is good. |
| 19:00.00 | *** join/#asterisk Amorsen (n=Amorsen@94.127.50.7) |
| 19:00.05 | citywok | the 480i is okay, the 6757i speaker is even better |
| 19:00.20 | citywok | i have a few 480i test units that are a bit unstable, the 57i never crashes like the 480i |
| 19:00.50 | [TK]D-Fender | I'll counter the "57i Not crashing" comment. Mine was Hatian. |
| 19:00.55 | ManxPower-work | I dislike Cisco's "Customer Hostile" policies when it comes to using their phones with 3rd party servers. |
| 19:00.59 | [TK]D-Fender | Haitian* |
| 19:01.10 | citywok | ooooooh, tooo sooooooon |
| 19:01.32 | citywok | that was pretty funny tk |
| 19:01.35 | ManxPower-work | We would have standardized on Cisco if they acted like they wanted our business. |
| 19:01.50 | citywok | but yea, i havent had any issues with mine crashing. i had 1 tank on a firmware update tho |
| 19:02.01 | Naikrovek | yeah they're arrogant assholes the cisco guys |
| 19:02.53 | highvoltz | I'm receiving a call into the asterisk box, however I get "number not in service" even though I can see the call hitting the box (using tail -f /var/log/asterisk/full) and the system is playing ss-noservice. What might be the cause of this? |
| 19:03.10 | citywok | higholtz watch the console, it will tell you why |
| 19:03.18 | citywok | likely you dont have the number added to a dialplan |
| 19:03.45 | p3nguin | highvoltz: Use the asterisk CLI "asterisk -r" and run "core set verbose 10" and watch what happens. Copy it all and put it into a pastebin so we can help you with it. |
| 19:04.20 | ruben23 | <PROTECTED> |
| 19:04.40 | p3nguin | Licenses are for CCM, not Asterisk. |
| 19:04.49 | Amorsen | ruben23: You need to get them onto the SIP firmware or run Skinny and cry |
| 19:05.04 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 19:05.15 | highvoltz | ok thax |
| 19:06.09 | jaskew | Bump: I'm cleaning up a page on voip-info.org and want to make sure I don;t do anything stupid... Has Free World Dialup passed into fond memory (i.e. are they dead)? |
| 19:06.16 | ruben23 | <PROTECTED> |
| 19:06.35 | Amorsen | ruben23: Phone-wise? We're fairly happy with Snom |
| 19:06.44 | [TK]D-Fender | highvoltz: GUI config error. NOT SUPPORTED HERE. Please return to #freepbx |
| 19:06.52 | Amorsen | Polycom DECT (former Kirk) is excellent |
| 19:06.55 | [TK]D-Fender | Polycom > ALL |
| 19:07.18 | *** join/#asterisk Tomae (n=Tomae@75-151-118-209-Washington.hfc.comcastbusiness.net) |
| 19:07.45 | Naikrovek | ^^^^ |
| 19:07.57 | Amorsen | Just avoid Grandstream and you'll be fine |
| 19:09.46 | citywok | http://www.google.com/search?q=free+world+dialup&ie=utf-8&oe=utf-8&aq=t&rls=org.mozilla:en-US:official&client=firefox-a |
| 19:11.44 | jameswf | I <3 Grand... nm cant do it |
| 19:11.55 | jaskew | citywok: I've been there ;) It seems to have died a while ago. SIP servers don;t respond. I want to make sure it isn't "just me" before I updarte the wiki |
| 19:12.28 | highvoltz | p3nguin, http://pastebin.com/d3ac12015 |
| 19:13.47 | Tomae | Is it safe to ask stupid questions? I am stumped and quite sure I am missing something simple. |
| 19:14.13 | highvoltz | I have one inbound route for any cid/did, to ring an ring group.. not sure what the problem is |
| 19:14.43 | voipmonk | Tomae: - SHOOT! |
| 19:14.52 | voipmonk | ask your question :) |
| 19:15.22 | voipmonk | highvoltz: is your inbound route utilizing sip? |
| 19:15.34 | ManxPower-work | highvoltz: This the second time you were told GUIs are not supported here. |
| 19:16.19 | highvoltz | ManxPower-work, asterisk related |
| 19:16.22 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 19:16.46 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 19:16.56 | [TK]D-Fender | highNO, that is GUI DIALPLAN not processing the call the way you WANT it to |
| 19:17.29 | [TK]D-Fender | highvoltz: the call makes it in. *'s job is done. |
| 19:18.30 | Tomae | Have a trixbox installation, newest versions of everything (as far as yum can do that), also did a full module update. I don't think that my issue is the config via the GUI as everything SEEMS to work. Incoming calls work via my SIP provider (Callcentric) and outgoing calls route and ring on the terminating end, but when one picks up the line there , there is silence. The CLI results look quite normal, however. |
| 19:19.01 | voipmonk | take that question - copy it and paste it into the #trixbox channel |
| 19:19.06 | Tomae | I even ditched the trixbox configuration, deleted it all, and did all my config via the aux config files. No joy. |
| 19:19.25 | voipmonk | but it looks like NAT |
| 19:19.49 | voipmonk | my tilda doesnt work anymore... |
| 19:19.53 | Tomae | Ahh, just figured as I also did it manually, I could ask here, as well. |
| 19:20.02 | Naikrovek | well |
| 19:20.09 | Naikrovek | there are other things going on in a trixbox system |
| 19:20.28 | Naikrovek | just because you're not using freepbx on trixbox anymore doesn't mean you've escaped trixbox entirely |
| 19:20.33 | Naikrovek | this is why trixbox sucks so much |
| 19:20.45 | ManxPower-work | highvoltz: very, very little in Asterisk GUIs is "asterisk". |
| 19:20.46 | voipmonk | ur gonna scare 'im 'way |
| 19:20.47 | Naikrovek | if you're comfortable with fiddling with the config files alone, i would do that |
| 19:20.58 | Tomae | Yeah... sadly I can only work with what I am provided, lol |
| 19:21.08 | Naikrovek | yeah same here |
| 19:21.15 | Naikrovek | i have a trixbox system and i'm stuck with it for the moment |
| 19:21.31 | voipmonk | till the next scheduled error |
| 19:21.36 | Naikrovek | yeah |
| 19:21.47 | Tomae | I just don;t understand how inbound can work PERFECTLY, but outbound is... sadfaced |
| 19:22.05 | Naikrovek | i want to get a parallel asterisk system going and then employ DUNDi to move extensions one at a time |
| 19:22.15 | [TK]D-Fender | voipmonk: If only it were that easy... |
| 19:22.27 | Katty | hi voipmonk |
| 19:22.33 | voipmonk | hello there |
| 19:23.13 | ManxPower-work | SIP call comes in, hits the GUI generated sip.conf. Call gets to the dialplan and hits the GUI generated extensions.conf. |
| 19:24.06 | ManxPower-work | Call goes out of the system, hits the GUI generated macros for dialing out. Someone calls an extension, the call goes thru the GUI generated macros for calling an extension. |
| 19:24.17 | ManxPower-work | I think you can see a pattern here. |
| 19:24.20 | [TK]D-Fender | ManxPower-work: I'd support at least getting the SIP peer to auth. Everything beyond it... FUKKIT |
| 19:24.23 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 19:24.52 | ManxPower-work | Tomae: we don't understand it either. |
| 19:24.55 | Naikrovek | i wish trixbox were better supported. it runs really nice until it breaks |
| 19:26.00 | russellb | it's great except for when it sucks? |
| 19:26.09 | Naikrovek | i really need to learn how to do asterisk without trixbox ... |
| 19:26.11 | Naikrovek | exactly |
| 19:26.15 | russellb | :-) |
| 19:26.50 | Naikrovek | it's a really well integrated system that works well. the ui is better than asterisknow (which i also like) but i want to not rely on those UIs in the future |
| 19:27.08 | Naikrovek | but when it breaks, it is very difficult to diagnose and to fix |
| 19:27.50 | Naikrovek | just need to make some time at work to learn the dialplan and contexts and security and that sort of thing |
| 19:28.04 | Tomae | I am used to doing it manually... |
| 19:28.12 | [TK]D-Fender | Naikrovek: Which AsteriskNOW? |
| 19:28.14 | Tomae | So, having to use trixbox is throwing me for a loop |
| 19:29.55 | *** join/#asterisk moy (n=moy@bas1-unionville55-1177733883.dsl.bell.ca) |
| 19:30.36 | *** join/#asterisk Tim_Toady (n=moi@193.92.60.66) |
| 19:31.03 | Katty | WAHHH |
| 19:31.14 | Katty | rips hair out |
| 19:31.46 | [TK]D-Fender | Katty: You go Brittney! |
| 19:32.37 | Katty | ^_- |
| 19:32.50 | Katty | your statement does not parse. |
| 19:32.59 | benngard | read ***Katty's tits out, i was thinking for a while... |
| 19:33.20 | Katty | not amused. |
| 19:34.09 | [TK]D-Fender | Katty: Does too....... |
| 19:34.31 | Katty | [TK]D-Fender: lies. |
| 19:34.37 | Katty | [TK]D-Fender: ALL LIES |
| 19:34.51 | [TK]D-Fender | Katty: YOU CAN'T HANDLE THE TRUTH! |
| 19:35.09 | Katty | you can't handle your face. |
| 19:35.23 | bmoraca_work | Naikrovek, elastix > trixbox...but it's still based on freepbx and as such is very hard to diagnose problems |
| 19:35.59 | Katty | i'm going to sneeze. |
| 19:36.11 | [TK]D-Fender | Naikrovek: #FreePBX -ers consider Elastix to be only slightly less evil than Trixbox. They recommend PIAF by comparison |
| 19:36.12 | Naikrovek | freepbx alone isn't so bad |
| 19:36.30 | Naikrovek | i want a gui-less system |
| 19:36.34 | [TK]D-Fender | Naikrovek: At least PAIF runs off mainstream FreePBX unlike the rest of the forkers |
| 19:36.43 | [TK]D-Fender | Naikrovek: Then by all means... |
| 19:36.48 | Naikrovek | never used PIAF or Elastix |
| 19:36.51 | Naikrovek | no desire to |
| 19:36.59 | Tomae | Charming. I set it up on the smartphone, and I get silence on outbound for that as well |
| 19:36.59 | Naikrovek | want to sit down and learn vanilla asterisk |
| 19:37.07 | citywok | Naikrovek: it's not difficult |
| 19:37.10 | ManxPower-work | I've spent the past month rewriting parts of FreePBX to make it suck less. |
| 19:37.15 | citywok | i went from no asterisk at all, to vanilla asterisk |
| 19:37.18 | Naikrovek | ManxPower-work: nice |
| 19:37.26 | Naikrovek | citywok: i know, i have the book, just need to go through it |
| 19:37.34 | citywok | to home brewing a fully integrated * solution for our call center |
| 19:37.35 | bmoraca_work | Elastix is mainstream freepbx. Elastix also has a lot of other stuff built in and GUIs for various other things in an attempt to make a unified-communications-in-a-box-for-dummies |
| 19:37.48 | ManxPower-work | Naikrovek: Not really "rewriting", but "writing from scratch, then interfacing it with FreePBX to augment or replace specific FreePBX things. |
| 19:37.51 | citywok | Naikrovek: i didn't even use a book, heh -- the most trouble i had was around T1 spans, and integrating with my closed-source Inter-Tel Axxess system |
| 19:38.06 | Naikrovek | ManxPower-work: ah even better |
| 19:38.17 | mlr78 | exit |
| 19:38.24 | citywok | dont forget the / |
| 19:38.30 | ManxPower-work | And really more "write from scratch" ask my boss to integrate into FreePBX |
| 19:38.44 | Naikrovek | citywok: i don't get contexts. dialplan syntax i'm fine with, i don't get contexts. and i know 6 people are going to explain it to me now, all at once, but that's what i haven't learned yet |
| 19:39.09 | ManxPower-work | Naikrovek: contexts may be the most important (and most difficult) thing in Asterisk to understand. |
| 19:39.10 | citywok | they are just different... directions/roads/streets |
| 19:39.25 | citywok | you just use them to route in different way and follow different dialplans |
| 19:39.27 | Naikrovek | well that's what I'm hung up on |
| 19:39.41 | Naikrovek | not hung up so much as I haven't had time to figure it out myself |
| 19:39.42 | Naikrovek | yet |
| 19:39.44 | citywok | idk, it was pretty easy to me to grasp so i'm not sure |
| 19:39.48 | bmoraca_work | Naikrovek, http://www.merriam-webster.com/dictionary/context ...in asterisk, they're as literal as the definition |
| 19:42.22 | p3nguin | naikrovek: A context is basically just a section of extensions.conf seperate from all other sections. |
| 19:42.41 | Naikrovek | yeah i know that, and a call can't cross the context boundary without an include |
| 19:42.44 | Naikrovek | ... i think |
| 19:42.49 | Naikrovek | i need to sit down and do it |
| 19:42.52 | Naikrovek | then i'll know what i don't know |
| 19:42.59 | p3nguin | or a Goto() or a GotoIf() or a ... |
| 19:43.15 | ManxPower-work | You'll never understand contexts by looking at a GUI generated dialplan. |
| 19:43.25 | Naikrovek | ManxPower-work: yeah, i don't bother looking at those |
| 19:43.39 | Naikrovek | i try to find small examples online and learn from those |
| 19:43.46 | Naikrovek | and i've sat with the book on the toilet a few times and read |
| 19:43.53 | Naikrovek | i probably know more than I think I do |
| 19:44.03 | p3nguin | That's my suspicion. |
| 19:44.08 | Naikrovek | but i won't have the confidence until I sit down and set it up, from scratch, myself. |
| 19:44.26 | Naikrovek | i need to set up a few servers and set up DUNDi on them |
| 19:44.33 | Katty | dances |
| 19:44.58 | Naikrovek | and i want extension configs to come from mysql so i won't have to replicate a lot of setup on each server |
| 19:45.06 | bmoraca_work | Naikrovek, DUNDi isn't as useful as it appears...there are some limitations to it that make it not quite useful in a lot of cases |
| 19:45.14 | Naikrovek | bmoraca_work: please explain |
| 19:45.16 | ManxPower-work | Naikrovek: get it working and designed before doing the complicated stuff |
| 19:45.19 | citywok | when my receptionist transfers a call to a person, instead of the callerid being that of the person calling in, it's the receptionist, is there a way to change that? |
| 19:45.29 | ManxPower-work | citywok: Yes. |
| 19:45.52 | citywok | is it a switch in the dial command? |
| 19:46.27 | *** join/#asterisk Micc (n=quassel@c-98-225-57-96.hsd1.wa.comcast.net) |
| 19:46.45 | ManxPower-work | citywok: 1) Use BLIND transfer or 2) see "core show application dial" the option should be obvious. |
| 19:47.18 | bmoraca_work | Naikrovek, DUNDi was designed somewhat like dynamic routing protocols, in that it's a distributed lookup for numbers. the problem is that you're still manually required to enter in all TNs associated with each peer in the DUNDi contexts...it's not a useful application if peers have more than one associated TN |
| 19:47.20 | citywok | gotcha, iw as reading the o command on voip-info |
| 19:47.37 | ManxPower-work | When you do a transfer the person receiving the transfer should know who they will be speaking to when they answer the call. In an ATTENDED/SUPERVISED transfer that person is the person making the transfer. In BLIND transfer, that person is the original caller. |
| 19:47.38 | bmoraca_work | which, in my opinion, is the best possible case for using DUNDi |
| 19:47.40 | citywok | will it display Reception first, then the other callers callerid, or will it always show the other callers callerid? |
| 19:47.46 | Micc | Is there a way to make Park actually park the call right away and not after the parking phone is hung up? |
| 19:47.50 | Naikrovek | bmoraca_work: what's a TN :) |
| 19:47.56 | lupine_85 | aha, so I have a dialplan rule that matches outgoing SIP (exten => _.@.,1,Dial(SIP/...) |
| 19:47.57 | bmoraca_work | Naikrovek, telephone number |
| 19:48.09 | citywok | ManxPower-work: agreed, but very often its handy to see the callers number on your display so you can write it down |
| 19:48.20 | lupine_85 | I just need to work out which variable contains the full sip address I'm trying to get to |
| 19:48.30 | *** join/#asterisk grEvenX (n=even@cC0FD00C3.dhcp.bluecom.no) |
| 19:48.33 | bmoraca_work | Micc, blind transfer? |
| 19:48.48 | ManxPower-work | lupine_85: They are all listed in a very secret document located at /path/to/src/doc/tex/channelvariables.tex |
| 19:48.50 | citywok | ManxPower-work: and probably 2/3 of the battle is going to be the fact that it doesnt act the exact same way as the old phone system, so it's different, so omgwtfbbq |
| 19:50.37 | Micc | bmoraca_work, oh, is that why its doing that? That makes sense. I just set up a key on an aastra to be spre 300 and it does an attended transfer to 300 I guess. |
| 19:50.45 | Naikrovek | bmoraca_work: ah. well for my purposes the multi-server system will have extensions with extension numbers and some DIDs |
| 19:50.55 | lupine_85 | ManxPower-work: sounds like a crazy place to put them ;) |
| 19:51.03 | lupine_85 | peeks |
| 19:51.27 | Micc | bmoraca, do you know how to setup an aastra phone to do a bind transfer with a programmable key? |
| 19:52.01 | citywok | Micc, i did this, 1 sec let me get my config |
| 19:52.05 | Naikrovek | why do you want to blind transfer to a parking lot? doesn't asterisk read the parked extension number back to you, and don't you need to hear that? |
| 19:52.15 | Naikrovek | or is that some trixbox thing i'm speaking of |
| 19:52.17 | bmoraca_work | Naikrovek, basically, you'd be better off using realtime to get the SIP address of the peer and using some custom dialplan and func_odbc to construct a dynamic dialplan to distribute the calls between servers (or if a server fails, to route around that server) |
| 19:52.48 | bmoraca_work | Naikrovek, you can use hints to monitor parking spaces...additionally, you can glean that information from AMI |
| 19:52.50 | Naikrovek | bmoraca_work: okay |
| 19:52.53 | Micc | Naikrovek, yes, but I have blf, I don't really need to hear it. |
| 19:53.06 | citywok | Micc, i messaged you my cfg |
| 19:53.09 | bmoraca_work | Micc, i don't know anything about Aastra phones :) |
| 19:53.26 | Poincare | anyone ever used such a GoIP device? |
| 19:55.18 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 19:55.24 | *** join/#asterisk angryuser_laptop (n=angryuse@90-156-167-83.reverse.alphalink.fr) |
| 19:55.33 | Micc | Can I blind xfer into a specific parking spot or do I have to let it assign the parking spot? I guess if it failed because the spot was full, then it would have to connect you back to the caller. I'm not sure how that would work on some phones. |
| 19:56.14 | Micc | It would be nice to have a one touch park though. |
| 19:56.31 | p3nguin | Check your features.conf |
| 19:56.46 | *** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk) |
| 19:58.41 | Micc | Anyone know of a way I can do presense on a flag in a database? |
| 19:58.56 | Micc | I want to use BLF to show the status of forwarding on or off. |
| 19:59.44 | ManxPower-work | Micc: you can in 1.6.x.x |
| 20:00.52 | *** join/#asterisk slinksh0t_ (n=slinksh0@74.115.208.59) |
| 20:01.39 | *** join/#asterisk j_kroon (n=jkroon@dsl-244-22-201.telkomadsl.co.za) |
| 20:02.29 | *** join/#asterisk jarrod (i=anon@69.172.129.29) |
| 20:03.00 | jarrod | hey, if asterisk is logging cdr to Master.csv and I rename that file and touch another... would there be a brief moment where a call record would not get logged? |
| 20:04.01 | ManxPower-work | jarrod: no. Calls will continue to be logged to the new name until you issue a logger reload command |
| 20:04.20 | ManxPower-work | then it will close the file and re-open it, at which time logging will go to the new file. |
| 20:04.24 | jarrod | is that because its pointed to an inode? |
| 20:04.38 | ManxPower-work | jarrod: Yes. |
| 20:04.41 | jarrod | because i have a program that renames Master.csv to <date>.csv and touches Master.csv |
| 20:04.46 | jarrod | and it keeps logging to Master.csv |
| 20:04.48 | *** join/#asterisk TommyBotten (i=tommy@91.205.185.140) |
| 20:04.56 | jarrod | to the new Master.csv (without reload) |
| 20:05.09 | ManxPower-work | jarrod: very odd. |
| 20:05.17 | jarrod | i thought so |
| 20:05.40 | jarrod | linux/ext3 asterisk 1.4.6 |
| 20:05.51 | jarrod | didnt make sense to me either |
| 20:06.02 | ManxPower-work | Are you sure you're not running some form of GUI or running "reload"? |
| 20:06.08 | jarrod | absolutely |
| 20:06.17 | jarrod | i wrote a c program that handles importing the csv to sql |
| 20:06.25 | jarrod | and its rename, and create |
| 20:06.52 | TommyBotten | Using a dahdi analog card (FXS) and an analog device, the DTMF digits 2 and 6 does not seem to be recognized. While loading the respective dahdi module, I get indications that the 'us' tone set is loaded as well as the norwegian set. Could this be the case? And how can I prevent the US set from being loaded? |
| 20:07.21 | ruben23 | guys one more thing, cisco IP phones model with like--Cisco CP-7940G cannot be used for asterisk..? is this true..? |
| 20:07.22 | ManxPower-work | TommyBotten: increase or decrease the rxgain for those channels |
| 20:07.42 | ruben23 | i got this link--->http://www.888voipstore.com/cisco-sccp-phones/ |
| 20:07.52 | Micc | ManxPower, I'm running 1.6.1.12, so where can I find out about how to do that blf thing? |
| 20:08.02 | *** join/#asterisk [Outcast] (n=anonymou@64.202.62.5) |
| 20:08.13 | ManxPower-work | Micc: it's called DEVSTATE (I think it's a function) |
| 20:08.20 | Micc | thanks |
| 20:08.21 | ManxPower-work | function names are CASE SENSITIVE |
| 20:08.35 | TommyBotten | ManxPower-work: Is this done in asterisk-specific config or dahdi-specific? |
| 20:08.36 | ManxPower-work | Micc: also see the UPGRADE*.txt files and "core show functions" and "core show applications" |
| 20:09.14 | ManxPower-work | TommyBotten: where ever rxgain appears in the sample configs that come with Asterisk, (I think it's /etc/dahdi/system.conf, but I'm not sure) |
| 20:09.37 | TommyBotten | ManxPower-work: Ok, I'll test that. Thanks for the help :) |
| 20:10.42 | ManxPower-work | TommyBotten: DTMF issues on analog is almost always a volume level issue (two low -- doesn't detect all digits or too high - digits are distorted or doubled because of echo caused by the high volume) |
| 20:10.56 | p3nguin | ruben23: That's totally not true at all. |
| 20:11.14 | ManxPower-work | also do NOT set relaxdtmf=yes It will cause strings like 55 to be detected as "5" |
| 20:11.20 | p3nguin | ruben23: Useragent : Cisco-CP7940G/8.0 |
| 20:11.23 | jarrod | manx |
| 20:11.30 | jarrod | asterisk reopens the file by name for each write |
| 20:11.33 | jarrod | that seems inefficient |
| 20:11.56 | ManxPower-work | jarrod: I didn't think it used to do that, but I'm not an expert on CDRs, as I have never billed for calls. |
| 20:12.06 | Tomae | I think I got it done chussed out |
| 20:12.22 | TommyBotten | ManxPower-work: Much appreciated. Interesting stuff :) |
| 20:12.22 | p3nguin | ruben23: You can either use the factory SCCP image on the phone and use asterisk's chan_skinny... or you can get a SIP image to put on the phone and use the more-often-used chan_sip. |
| 20:12.24 | ManxPower-work | (When you're paying 30K/yr to work at a company, we included free calling) |
| 20:13.02 | p3nguin | You have to pay to work at a company? |
| 20:13.18 | ManxPower-work | p3nguin: Real Estate. It's called a "desk fee" |
| 20:13.20 | Tomae | I am just going to have to wait until the ports are forwarded |
| 20:13.25 | p3nguin | I want THAT business model. |
| 20:13.31 | p3nguin | I could be rich! |
| 20:14.17 | ManxPower-work | Like a waiter at an upscale restaurant may pay the restaurant to work there -- they get plenty in tips to cover it. |
| 20:15.13 | p3nguin | I'm looking to hire a couple consultants... I bet if I put in the ad that they have to PAY ME to work for me, I won't get as much response as if I offer to pay them. |
| 20:15.36 | p3nguin | I could be wrong. |
| 20:15.38 | ManxPower-work | p3nguin: If they make enough money on referrals they may not have a problem with it. |
| 20:15.42 | Micc | ManxPower, how do I set the initial DEVICE_STATE of a custom field? |
| 20:15.42 | TommyBotten | p3nguin: Yeah, it's a strange thing |
| 20:15.58 | p3nguin | I'm truely intrigued by this concept. |
| 20:16.00 | ManxPower-work | In real estate selling 1 house could cover your entire desk fee for the month. |
| 20:16.13 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:16.42 | ManxPower-work | Or selling one commercial property could cover your desk fees for an entire year. |
| 20:19.28 | *** join/#asterisk lanning (n=lanning@208.87.235.224) |
| 20:20.27 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 20:21.53 | *** join/#asterisk socain (n=socain00@74.255.249.66) |
| 20:21.57 | TheDavidFactor | what is the advantage of using trunking with iax? |
| 20:22.08 | *** part/#asterisk socain (n=socain00@74.255.249.66) |
| 20:22.52 | TommyBotten | TheDavidFactor: Easier when it comes to NATing/firewalling and less overhead |
| 20:23.05 | TommyBotten | TheDavidFactor: Overhead in the sense of traffic |
| 20:23.07 | khronos | You also get a bandwidht savings if it works out for you. |
| 20:23.24 | khronos | The more calls you roll at one time the more you save over sip. |
| 20:23.35 | TheDavidFactor | so maybe not worth it if I've got two asterisk servers sitting on the same network segment that are talking to each other? |
| 20:24.03 | khronos | I've really only used iax trunking on at most 8 calls at once so not really sure how it scales though. |
| 20:24.22 | TommyBotten | I've run some twenty calls without trouble |
| 20:24.30 | TheDavidFactor | cool, thanks! |
| 20:25.29 | ManxPower-work | iax2 trunking has nothing to do with NAT |
| 20:25.34 | *** join/#asterisk [Outcast] (n=anonymou@64.202.62.5) |
| 20:25.59 | *** join/#asterisk Mango (n=iMango@d154-20-97-118.bchsia.telus.net) |
| 20:26.18 | ManxPower-work | TheDavidFactor: IAX2 trunking can save significant amounts of bandwidth when you have 3 or more calls between the same two Asterisk servers. That is all it does. |
| 20:26.25 | p3nguin | IAX2, on the other hand, is often easier to pass through NAT than SIP/RTP. |
| 20:26.36 | ManxPower-work | p3nguin: only if you don't know what you are doing. |
| 20:26.49 | p3nguin | Or when the networking appliances suck. |
| 20:27.12 | ManxPower-work | Is it really that much harder to add 2 lines to sip.conf and port forward multiple ports instead of one port. |
| 20:27.15 | *** join/#asterisk dennis00 (n=Dennis_A@unaffiliated/dennis00) |
| 20:27.23 | dennis00 | pbx.c: Timeout, but no rule 't' in context 'phones' < What does this T mean? |
| 20:27.34 | p3nguin | You shouldn't have to forward any ports on the client's side. |
| 20:27.35 | ManxPower-work | You still need to port forward if your Asterisk server is behind NAT when using IAX2 |
| 20:27.37 | TheDavidFactor | ManxPower-work, thanks! |
| 20:27.58 | ManxPower-work | dennis00: it means you have an IVR that is waiting for input. No input was received so the dialplan will jump to exten => t |
| 20:27.58 | *** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca) |
| 20:28.12 | ManxPower-work | dennis00: and is it "t" or "T" They are different |
| 20:28.16 | p3nguin | When the client's networking appliances suck, that is when IAX2 is better than SIP/RTP over NAT. |
| 20:28.26 | dennis00 | ManxPower-work: t |
| 20:28.28 | ManxPower-work | p3nguin: yes dear. |
| 20:28.39 | carrar | get better clients |
| 20:28.55 | ManxPower-work | What networking appliance lets you forward 1 port and not let you forward more than 1 port? |
| 20:29.19 | p3nguin | We're talking about client side, still, which you shouldn't be forwarding any ports on in the first place. |
| 20:29.20 | ManxPower-work | dennis00: you really should read the Asterisk book |
| 20:30.26 | ManxPower-work | I'd just go to Wal-Mart and get a $60 Linksys router for those clients. |
| 20:31.00 | *** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1) |
| 20:31.17 | dennis00 | We dont have Walmarts. |
| 20:31.25 | ManxPower-work | Better than "My car sucks so I'll just drive very slow". If your car sucks that much replace the car. |
| 20:31.38 | ManxPower-work | dennis00: do you have postal service? |
| 20:31.39 | leifmadsen | or don't have a car |
| 20:32.05 | *** join/#asterisk Buklov (n=buklov@213.138.71.254) |
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| 20:34.01 | dennis00 | ManxPower-work: Yes, but expensive. |
| 20:34.17 | ManxPower-work | Is it more expensive than having an unhappy customer? |
| 20:34.30 | dennis00 | ManxPower-work: I dont sell VOIP and I am not planning to. |
| 20:36.57 | p3nguin | For a system that is already set up on a PRI, and the customer wants to add fax support, what type of card would you recommend to add to the box to add analog fax capability? |
| 20:38.08 | dennis00 | I would probably ask you. |
| 20:38.35 | Katty | :< |
| 20:38.54 | Katty | why do i always have to pee at the most inconvienent times. |
| 20:39.01 | Katty | like while on the phone with 3 people >.< |
| 20:39.15 | p3nguin | They already have a Digium card for the PRI, so I'm sure they will be willing to get another Digium card for fax ports. |
| 20:39.30 | p3nguin | Just don't know what model to say to get. |
| 20:39.43 | Katty | fax ports? |
| 20:39.49 | Katty | ^_- |
| 20:39.54 | *** join/#asterisk Ad-Hoc (n=nimbus@62.1.172.179.dsl.dyn.forthnet.gr) |
| 20:40.35 | ChannelZ | anal-log? |
| 20:40.49 | dennis00 | pardon me? :P |
| 20:42.05 | Katty | fax machines. |
| 20:42.08 | Katty | get one. |
| 20:42.47 | dennis00 | I used a PAP2T adaptor for that purpose, but I was not controlling the server-end. |
| 20:42.52 | p3nguin | I assume they already have a fax machine or they wouldn't be asking for a card to add to the machine. |
| 20:43.06 | dennis00 | I lost my power adapter for it, it costs $30 to get it here. :/ |
| 20:43.25 | Katty | p3nguin: that statement doesn't parse. |
| 20:43.30 | Katty | p3nguin: and i re-read it 3 times |
| 20:43.38 | Katty | checks drink for suspicious additives. |
| 20:43.46 | p3nguin | Which part are you having trouble with? |
| 20:44.53 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 20:45.14 | Katty | if they already had a fax machine |
| 20:45.18 | bmoraca_work | p3nguin, you'll have much better luck with an ATA than with an analog digium card |
| 20:45.22 | Katty | why would they want asterisk to do faxing, period? |
| 20:45.40 | Katty | i don't get it. |
| 20:45.49 | dennis00 | so they dont need a landline that costs $20 monthly? |
| 20:46.04 | bmoraca_work | actually, let me rephrase that...you won't need luck with an ATA, as it'll work just fine. you will need luck with a Digium analog card, however |
| 20:46.23 | p3nguin | katty: Well, they have a PRI with 50 spare DIDs which goes into an Asterisk box via Digium PRI card. |
| 20:46.26 | dlynes | dennis00, it's only $20/mo for a business fax line where you live? damn...I wish it was somewhere close to that here |
| 20:46.52 | p3nguin | katty: From there, they wish to add the ability to connect a fax machine so they can utilize at least one of the spare DIDs for a fax number. |
| 20:46.57 | *** join/#asterisk nicknick (n=nicholas@93-97-188-195.zone5.bethere.co.uk) |
| 20:47.04 | dlynes | bmoraca_work, really? it works just fine with a sangoma pri card and a sangoma analog card |
| 20:47.06 | bmoraca_work | Katty, i've always set it up such that faxes are received by asterisk (or, more precise, hylafax) and then they have an ATA with a fax machine attached for outbound |
| 20:47.08 | Katty | i've never seen that before. |
| 20:47.24 | Katty | i've just seen faxes routed to a user's email address via DID |
| 20:47.26 | bmoraca_work | dlynes, yes, it usually works, and i've set it up as such before...but ATAs are MUCH cheaper and MUCH more reliable for faxing |
| 20:47.35 | Katty | for that, asterisk okay. for sending faxes..... |
| 20:47.39 | Katty | go buy a fax machine. |
| 20:47.42 | Katty | a REAL fax machine. |
| 20:48.06 | dlynes | bmoraca_work, ah...i've had zero problems with that configuration...the bridging happens and it just .... works |
| 20:48.11 | p3nguin | bmoraca_work: So you would recommend using SIP between * and the ATA? |
| 20:48.12 | dlynes | bmoraca_work, no fuss |
| 20:48.13 | dennis00 | My recent calls in voipbuster says -0.01 ct but my saldo says 9.90, I dont trust it. |
| 20:48.47 | dlynes | bmoraca_work, yet, i've had all kinds of problems with sip and fax...even over a 10' network cable |
| 20:49.07 | bmoraca_work | p3nguin, yes...use a PAP2T with g.711 (or T.38 if 1.6) and you won't have any problems |
| 20:49.22 | p3nguin | Sounds like a good plan to me. |
| 20:49.45 | p3nguin | This is a better idea than adding any type of analog card... even the best of the best? |
| 20:49.46 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 20:49.52 | bmoraca_work | dlynes, i've faxed over SIP over a 100ms latency connection with roughly 30 hops. works every time. |
| 20:50.02 | bmoraca_work | p3nguin, i've never had a problem doing it this way |
| 20:50.06 | ManxPower-work | p3nguin: with an ATA even if it does not work, it didn't cost you very much |
| 20:50.16 | bmoraca_work | that too |
| 20:50.17 | p3nguin | Very good point you have. |
| 20:51.05 | ManxPower-work | It's always good to have a spare ATA or two as well. |
| 20:51.43 | bmoraca_work | i generally still suggest that my customers keep an analog line, anyway, as i don't want to be liable for things like alarm lines and emergencies when internet is out...but that's just me |
| 20:51.48 | dlynes | bmoraca_work, using t.38? or straight ulaw? |
| 20:51.53 | dennis00 | or a spare power adaptor if you only have 1 ATA :( |
| 20:51.53 | bmoraca_work | dlynes, straight ulaw |
| 20:52.12 | p3nguin | bmoraca_work: So with the ATA/Fax method, asterisk still needs to receive the fax (using hylafax) instead of just routing the fax call to the ATA? |
| 20:52.19 | dlynes | bmoraca_work, I don't understand how all you guys leading charmed lives are able to pull that ulaw faxing off |
| 20:52.37 | bmoraca_work | p3nguin, no, no, no...just have asterisk route the call straight to the SIP peer on the ATA |
| 20:52.39 | dlynes | bmoraca_work, you must have a dozen horseshoes jammed up your ass, or something |
| 20:52.45 | *** join/#asterisk evil_gordita (n=evilgord@ip70-188-50-186.rn.hr.cox.net) |
| 20:52.57 | p3nguin | bmoraca_work: Okay, good. I guess I misunderstood what you were saying about hylafax. |
| 20:53.00 | bmoraca_work | dlynes, lol...i could say the same about you and not having problems with an analog card |
| 20:53.11 | p3nguin | You were having asterisk keep the faxes that came in, I guess. |
| 20:53.14 | bmoraca_work | p3nguin, that was a different implementation option...not necessary |
| 20:53.21 | lupine_85 | ok.... it /looks/ like I've got a dialplan that lets me make peer-to-peer SIP calls. I send the SIP Invite with the correct From and To headers set, then I get an ?invite? packet from the peer I just sent an invite to and my asterisk responds 401 unauthorized.... |
| 20:53.23 | p3nguin | The fax machine and ATA were only for outgoing faxes. |
| 20:53.31 | dennis00 | Do people here have experience with Asterfax? |
| 20:53.44 | bmoraca_work | dlynes, although i did once get a 44kbps dialup connection over a sangoma analog card through asterisk...that was fun |
| 20:53.59 | dlynes | dennis00, the odd person out of everyone that does fax on asterisk, so probably 0.001% of the people on this channel |
| 20:54.02 | bmoraca_work | p3nguin, correct. it's but one option for implementing fax in an office |
| 20:54.21 | dennis00 | dlynes: I see. :) |
| 20:54.38 | dlynes | dennis00, and even that's being generous |
| 20:55.25 | bmoraca_work | mostly, i try to keep data out of the PBX...and that includes faxing and CC machines |
| 20:55.55 | bmoraca_work | i haven't checked lately, but has anyone ported nvfaxdetect to 1.6 yet? |
| 20:58.11 | *** join/#asterisk Geminizer (n=whoami@cpe-76-180-27-4.buffalo.res.rr.com) |
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| 20:59.09 | Geminizer | Has anyone ever had an issue where they invoke Dial with a given timeout value and the call hangs up way earlier than the timeout specified? |
| 20:59.31 | bmoraca_work | Geminizer, what type of trunk? |
| 20:59.40 | Geminizer | SIP |
| 20:59.52 | p3nguin | bmoraca_work: What if they want 4-8 analog ports? Still recommend a bunch of ATAs? Apparently the building has 5 companies all sharing the PRI. |
| 21:00.28 | bmoraca_work | p3nguin, get an ATA with 4-8 ports. the Cisco SPA8000 works pretty well (and has an amphenol connector, too) |
| 21:00.32 | carrar | p3nguin, use a 8 port audiocodes or a T1 Channelbank |
| 21:00.35 | p3nguin | I guess five ATAs is about $100, where a good analog card is much higher. |
| 21:00.52 | carrar | I'd go with a T1 channel bank |
| 21:01.11 | p3nguin | The SPA8000 allows for 8 different voip users? |
| 21:01.12 | bmoraca_work | carrar, way expensive, though, when you factor in the cost of the T1 card |
| 21:01.19 | carrar | $200 |
| 21:01.27 | bmoraca_work | p3nguin, yes. configuration is nearly identical to the PAP2T |
| 21:01.28 | carrar | for the channelbank |
| 21:01.40 | carrar | and way more reliable |
| 21:01.47 | bmoraca_work | carrar, link? |
| 21:01.51 | carrar | ADIT600 |
| 21:01.55 | carrar | get em used on ebay |
| 21:02.05 | jarrod | adit600 is junk |
| 21:02.09 | carrar | or find one on vacated office walls |
| 21:02.09 | [TK]D-Fender | carrar: as long as you don't care about warranty |
| 21:02.13 | jarrod | ill sell you one |
| 21:02.15 | carrar | ADIT600 works awesome |
| 21:02.19 | carrar | it is not junk |
| 21:02.25 | jarrod | i have a few ill sell you |
| 21:02.25 | carrar | it's solid |
| 21:02.33 | carrar | perhaps you broke them |
| 21:02.39 | jarrod | heh... no they are awesome |
| 21:02.40 | jarrod | solid |
| 21:02.55 | carrar | Thats also why major carriers use them |
| 21:02.59 | bmoraca_work | carrar, you still need the $400 T1 card, though. |
| 21:03.05 | carrar | because they are work horses |
| 21:03.34 | carrar | yeah, well if you are doing data on analog ports you alreadyhave a T1 card |
| 21:03.39 | carrar | just get additional port |
| 21:03.59 | bmoraca_work | which is still way more expensive than an SPA8000 |
| 21:04.25 | carrar | I build solid setups |
| 21:04.36 | bmoraca_work | a channel bank would be appropriate if you need many, many ports...but for that few, an ATA is better, imo |
| 21:04.36 | carrar | not spa |
| 21:04.44 | bmoraca_work | ok, so use an Adtran TA908 |
| 21:05.09 | carrar | Adtran is ok too, but you can find the ADIT's for cheap |
| 21:05.12 | p3nguin | They already have a Digium PRI card where their T1 connects to * |
| 21:05.26 | carrar | upgrade them to a 2 port |
| 21:05.38 | p3nguin | They just wanted to add fax machines to soak up a few of the unused DIDs they have. |
| 21:05.52 | carrar | I not put that over IP |
| 21:05.57 | carrar | I would not put that over IP that is |
| 21:06.58 | carrar | also the ADIT's come in groups of 8 channels at a time |
| 21:07.07 | Naikrovek | eyeballs the "Know your mate" section of the company newsletter... |
| 21:07.15 | carrar | so if you only need 8 channels you just get 1 FXS blade |
| 21:08.07 | carrar | I have customers who have been on them for years |
| 21:08.11 | p3nguin | Is it cheaper than a $250 SPA-8000? |
| 21:08.37 | bmoraca_work | p3nguin, not when you consider that you need a second T1 port, no. |
| 21:08.58 | carrar | shop around |
| 21:09.12 | bmoraca_work | is it a better solution? probably, to some extent. but it's much more expensive, and at this level, not needed. |
| 21:09.29 | carrar | SPA-8000 is IP |
| 21:09.47 | bmoraca_work | and faxing to them works just fine |
| 21:09.51 | carrar | build it cheap, well thats your call |
| 21:11.49 | carrar | You have the ability to keep those analog lines from never touching IP |
| 21:12.03 | carrar | I would use that ability |
| 21:13.04 | carrar | IP asin IP over your network |
| 21:13.06 | bmoraca_work | IP isn't the issue...the issue is Asterisk's treatment of the channels and limitations of its DSPs and other hardware limitations...that exists whether you're going IP or not. native bridging helps a bit, but doesn't fix the problem. |
| 21:13.36 | carrar | I've never had a problem with them |
| 21:13.49 | carrar | IP is typically the problem |
| 21:13.53 | bmoraca_work | just like i've never had a problem with ATAs. |
| 21:14.25 | p3nguin | I've recommended the SPA-8000. We'll see what happens. |
| 21:14.55 | bmoraca_work | for a lot of people, cost is the absolute dictator...if you tell them it's an extra $1000 to set it up, they're going to say no. |
| 21:15.31 | *** join/#asterisk uqlev (n=yuriy@91.184.221.31) |
| 21:16.19 | [TK]D-Fender | p3nguin: for what use? |
| 21:16.34 | carrar | fax |
| 21:16.39 | [TK]D-Fender | :~ |
| 21:17.24 | [TK]D-Fender | p3nguin: Whats your origin channel coming in on? |
| 21:18.14 | carrar | <p3nguin> They already have a Digium PRI card where their T1 connects to * |
| 21:18.19 | p3nguin | They have a PRI going to a Digium card in their * system. They have lots of spare DIDs, so they wanted to pipe a couple out to fax machines. |
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| 21:28.13 | carrar | ADIT 600 on ebay for $49 |
| 21:28.26 | carrar | 8 FXS ports |
| 21:28.32 | carrar | TDm controler and pwr |
| 21:29.00 | carrar | 35 mins left |
| 21:29.28 | [TK]D-Fender | checkout time, later all |
| 21:35.03 | ManxPower-work | I would go for Adtran TA750 or TA850 if you want a channel bank |
| 21:37.07 | *** join/#asterisk voipmonk (n=shido6@dsl-67-204-37-228.acanac.net) |
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| 22:14.14 | sbrath | anyone see a problem with a TE110P and a TDM400P in the same computer? When I put the TDM400P card in, the T1 card wouldn't sync with the uplink anymore... |
| 22:15.58 | *** join/#asterisk ashd (n=ashleyd@94-194-208-216.zone8.bethere.co.uk) |
| 22:23.46 | etnos | can someone please shed some light, I'm trying to set up conference calling but I get " you have entered an invalid conference number" and the phone call ends.. this is my meetme.conf and extensions.conf http://pastebin.com/d4c3ab196 what am i doing wrong? |
| 22:25.41 | p3nguin | Provide a valid conference number in your MeetMe() command or create one dynamically. |
| 22:26.05 | sbrath | MeetMe(${EXTEN},MsIqx) |
| 22:26.18 | *** join/#asterisk kaseano (n=kaseano@ip70-173-213-227.lv.lv.cox.net) |
| 22:26.20 | p3nguin | 'd' -- dynamically add conference |
| 22:26.42 | p3nguin | Either create the conference number ahead of time or create it dynamically. |
| 22:27.21 | kaseano | hey I'm trying to get a # to route to my box & set up w asterisk |
| 22:27.31 | kaseano | do I need a "virtual pri" or can I just get a voip # |
| 22:28.31 | p3nguin | That depends entirely on what you're doing. |
| 22:28.37 | carrar | ~itsp |
| 22:28.38 | infobot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 22:29.36 | *** join/#asterisk xpot-mobile (n=xpot@173-14-232-121-Utah.hfc.comcastbusiness.net) |
| 22:29.36 | leifmadsen | here's a tip: don't try and reproduce a problem with packet2packet bridging on asterisk 1.4.x |
| 22:29.42 | leifmadsen | (it doesn't exist in 1.4) |
| 22:31.28 | kaseano | p3nguin if I'm having multiple people calling in and getting forwarded/routed to other phone #s, do I need the virtual pri? |
| 22:31.44 | kaseano | or just putting people on hold til a line is available |
| 22:32.22 | p3nguin | I don't even know what a virtual PRI is. Anyone I ever asked doesn't know, either. |
| 22:32.30 | p3nguin | I know what a real PRI is. |
| 22:32.56 | kaseano | well I'll take that as I don't need 1 lol |
| 22:33.16 | p3nguin | If you want real voice lines, get a real PRI. |
| 22:33.30 | p3nguin | If you want straigh VoIP, you can get service from an ITSP. |
| 22:33.46 | kaseano | ok yeah that's what I did =) awesome thx p3nguin |
| 22:34.12 | *** join/#asterisk JAMMAN2110 (n=james@unaffiliated/jamman2110) |
| 22:34.15 | leifmadsen | p3nguin: virtual PRI I presume is just 23 (or 30) VoIP connections |
| 22:34.30 | p3nguin | If you are having a hard time getting to all the phone calls that are oming in, learn how to use Queue(). |
| 22:34.50 | leifmadsen | Queue() works awesome -- I've made a business off its use :) |
| 22:35.03 | kaseano | ok |
| 22:35.04 | p3nguin | If you need help with Queue() and queue.conf, just say what you're having trouble with. |
| 22:35.16 | Mango | leifmadsen: Funny thing is VoIP.ms allows 25 connections on their PAYG DIDs but charge extra for Virtual PRI. |
| 22:35.24 | Mango | I still don't know what it is either :( |
| 22:35.38 | leifmadsen | Mango: ya, virtual PRI doesn't compute |
| 22:35.56 | leifmadsen | p3nguin: I also added "Building a Queue" documentation to 1.6.2 and trunk the other day -- check the doc/ directory for more information |
| 22:36.09 | carrar | use the vitual internet |
| 22:36.15 | carrar | virtual |
| 22:36.22 | leifmadsen | I listen to virtual music every day |
| 22:36.24 | p3nguin | leifmadsen: Is there any way to pad a queue, meaning that instead of saying "you are now first in line," it would say "You are now 11th in line," or something like that? |
| 22:36.32 | carrar | heh |
| 22:36.45 | leifmadsen | p3nguin: uhhhh... I can't remember if that functionality exists -- I've never looked for that to be honest :) |
| 22:36.55 | leifmadsen | i.e. I've never had a customer request it |
| 22:37.27 | *** join/#asterisk zamba (i=marius@flage.org) |
| 22:37.34 | zamba | how can i force my sip clients to reregister? |
| 22:37.35 | p3nguin | I've had a person ask why it took 10 minutes when he was first in line. I was thinking if it was padded, he wouldn't ask. |
| 22:37.41 | p3nguin | zamba: sip reload |
| 22:38.01 | leifmadsen | zamba: reboot them? |
| 22:38.07 | zamba | leifmadsen: they are remote |
| 22:38.24 | zamba | and i've moved the asterisk server |
| 22:38.34 | p3nguin | run sip reload and then wait for the remote clients to timeout and register again. |
| 22:38.35 | zamba | updated the dns records.. and now waiting for the clients to merge over |
| 22:38.39 | leifmadsen | zamba: drop your max re-registration time in sip.conf, although not sure if that will send anything requesting them to re-register |
| 22:38.54 | zamba | leifmadsen: ok, i'll probably just have to wait for it to happen naturally, then |
| 22:38.59 | zamba | it's night time anyway :) |
| 22:39.10 | leifmadsen | p3nguin: hmmm, if they are actually 1st in line, they shouldn't be waiting 10 minutes :) |
| 22:39.32 | p3nguin | People like to talk. I can't just hang up. |
| 22:39.56 | p3nguin | If people didn't like to talk, we wouldn't be in this business. |
| 22:40.51 | paulc | smirks at p3nguin - us too (but almost certainly a different kind of business) |
| 22:41.12 | p3nguin | astersex? |
| 22:42.24 | sevv | dogballs |
| 22:43.23 | Qwell | p3nguin: disable the playing of position |
| 22:44.04 | p3nguin | I would rather leave it and be able to pad it slightly. But an excellent suggestion you have made. |
| 22:44.17 | random_mike | Can anyone advise if using Asterisk::AGI in perl, or perhaps if asterisk itself is able to read the privacy=full tag in the From field? IE: From: 1234 <sip:1234@xxx.xxx.xxx.xxx:5060>;tag=1e55bcd930;privacy=full |
| 22:49.50 | raden_work | when people call in some say they dial my extension 103 and it just loops back through the main menu anyone have a idea why this would be ? |
| 22:50.12 | p3nguin | dialplan fail |
| 22:50.36 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 22:50.46 | *** join/#asterisk vitaminx (n=vitaminx@89.130.31.1) |
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| 22:52.54 | Qwell | p3nguin: or for more fun, randomly rename the h-1 through h-9 sound files |
| 22:53.11 | p3nguin | hahaha |
| 22:53.25 | Qwell | or! |
| 22:53.30 | Qwell | just replace h-1 through h-3 with h-4 |
| 22:53.41 | Qwell | that way, everybody is at best the 4th caller |
| 22:53.53 | p3nguin | That could actually work as the padding I was looking for. |
| 22:54.11 | Qwell | heh |
| 22:56.05 | bmoraca_work | lol...random would be more fun |
| 23:02.37 | drfreeze | I've just setup an office with polycom phones and am getting a flashing date/time with the wrong date, time |
| 23:02.58 | drfreeze | Anyone know where I should be looking to fix this problem |
| 23:03.18 | outtolunc | VCR Setup 101 ? |
| 23:03.48 | sevv | haha. |
| 23:04.02 | paulc | drfreeze: NTP time server - set it to a good source in your office (fileserver that serves time, or maybe router) or go with one of the pool.ntp.org addresses |
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| 23:08.23 | p3nguin | qwell: I tried it, but it didn't seem to change what sound was played. |
| 23:08.32 | p3nguin | I reloaded app_queue, too. |
| 23:10.14 | random_mike | netsplit! |
| 23:10.19 | random_mike | run for your lives |
| 23:11.11 | p3nguin | ah, Playing 'queue-youarenext' |
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| 23:15.53 | etnos | p3nguin: thanks for the info, I tried it but no luck. I get this error dont know if it has anything to do with it: build_conf: Unable to open pseudo device |
| 23:16.13 | drfreeze | paulc: yeah, I set it in the dhcpd conf file: option ntp-servers 10.123.50.1; |
| 23:16.33 | drfreeze | but still get flashing. Also, I set the SNTP on the phone (is that NTP)? |
| 23:16.34 | p3nguin | etnos: Does MeetMe() work at all? |
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| 23:17.15 | etnos | p3nguin: nope, I get " you have entered an invalid conference number" |
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| 23:17.37 | p3nguin | etnos: Then you need to specify a valid conference number or create it dynamically. |
| 23:18.30 | p3nguin | meetme not working AT ALL is completely different than not knowing how to specify or create a conference. |
| 23:18.38 | etnos | p3nguin: I specified it dynamically like this: MeetMe(800,dM) |
| 23:19.13 | p3nguin | And calling to the exten that runs that gives the error about a pseudo device? |
| 23:19.29 | etnos | p3nguin: yeah |
| 23:19.51 | p3nguin | Then meetme isn't working. |
| 23:20.01 | p3nguin | ~meetme |
| 23:20.01 | infobot | methinks meetme is a conference bridge. More info at http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe |
| 23:20.36 | p3nguin | I use this: MeetMe(${EXTEN},dIMops) |
| 23:20.42 | p3nguin | It works fine. |
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| 23:22.15 | etnos | p3nguin: what about your meetme.conf? |
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| 23:22.44 | etnos | I only have conf=> 800 does that look fine? |
| 23:23.24 | p3nguin | If you have specified 800, then you don't need to create 800 dynamically in the MeetMe() command. |
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| 23:24.46 | p3nguin | I don't specify any because I don't use enough to care to list them. |
| 23:25.30 | etnos | p3nguin: so let me remove conf=>800 and see if it works |
| 23:25.51 | p3nguin | I can't imagine how that will magically fix meetme. |
| 23:26.04 | p3nguin | It seems like you didn't read the instructions to get it working. |
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| 23:44.17 | random_mike | Can anyone advise if using Asterisk::AGI in perl, or perhaps if asterisk itself is able to read the privacy=full tag in the From field? IE: From: 1234 <sip:1234@xxx.xxx.xxx.xxx:5060>;tag=1e55bcd930;privacy=full |
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| 23:56.03 | MmixX | hi all |
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| 23:56.16 | etnos | p3nguin: you are right I didnt. I will read more, thanks for your input |
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