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| 01:06.35 | rhollan | anyone have asterisk working with festival? |
| 01:07.02 | rhollan | I tried to configure it, and festival gets a request, but asterisk barfs on a non-zero exit on the spawn |
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| 01:49.52 | leifmadsen | what's that application that lets you send SIP messages to end points.... |
| 01:49.56 | leifmadsen | not SIPp.... |
| 01:49.58 | leifmadsen | the OTHER one :) |
| 01:50.10 | ChannelZ | sipsak? |
| 01:50.24 | leifmadsen | that's the wone |
| 01:50.24 | ChannelZ | which always sounds dirty to me |
| 01:50.25 | leifmadsen | one* |
| 01:50.29 | leifmadsen | lol |
| 01:50.30 | leifmadsen | thx :) |
| 01:50.39 | ChannelZ | heh sure |
| 01:51.13 | rhollan | How can I have Asterisk play MULTIPLE files after someone answers? A() in Dial only supports one file and I can't have multiple A() subcommands |
| 01:53.05 | p3nguin | You tried stringing several files together with the ampersand? |
| 01:53.11 | rhollan | To be clear, I am dialong, and then want to playback on answer. |
| 01:53.23 | rhollan | p3nguin: yup: took it as one big file name |
| 01:53.35 | leifmadsen | rhollan: if that doesn't work (try that first), then you could try the U() (or M() ) flags to execute dialplan so you can use Playback() |
| 01:53.44 | rhollan | File /var/lib/asterisk/sounds/Greeting&/tmp/festival does not exist in any format |
| 01:53.54 | leifmadsen | alternatively, use sox to concatenate the files together as a single file then use A() |
| 01:53.54 | rhollan | so & was interpreted as part fo the file name |
| 01:54.21 | rhollan | leifmadsen: yeah.... I'd like to avoid concatenating files if I can... seems kludgy. |
| 01:54.37 | leifmadsen | rhollan: see first suggestion |
| 01:55.03 | leifmadsen | goes to get out of his golf clothes |
| 01:55.10 | rhollan | The app has someone dial in, select an extension, causing dial out, and when the new party answers I want to play messages before connecting the two together: a general greeting and an anouncement of who called. |
| 01:55.29 | rhollan | looks at U() and M() |
| 01:55.45 | Kobaz | use U if you're on 1.6 + |
| 01:56.46 | Kobaz | wants an asyncronous U |
| 01:57.09 | rhollan | I'm on 1.6 (built for Ubuntu) but am considering moving to 1.8 |
| 01:58.05 | rhollan | Oh... M()... I read some bad stuff about macros. Is that why U() is preferred |
| 02:06.01 | rhollan | nope: U() did not work: After called party answered, parties were simply connected, no gosub, no nothing. |
| 02:08.23 | Kobaz | what's your dialplan look like |
| 02:08.26 | Kobaz | and did you reload |
| 02:08.40 | Kobaz | and does the context you specified exist? |
| 02:08.44 | rhollan | wait a sec, I coded W instead of U... and yes, I did reload :-) |
| 02:08.52 | rhollan | lemmy try again |
| 02:10.54 | rhollan | Attempt to reach a non-existent destination for gosub: (Context:called-greeting, Extension:s, Priority:1) |
| 02:10.54 | rhollan | must have the syntax to U() wrong. |
| 02:11.59 | rhollan | exten => 666,n,Dial(SIP/xxxxxxxxxx@sip.broadvoice.com,30,U(called-greeting)) |
| 02:12.00 | rhollan | exten => 666,n,Hangup() |
| 02:12.00 | rhollan | exten => 666,n(called-greeting),Playback(/var/lib/asterisk/sounds/Greeting) |
| 02:12.00 | rhollan | exten => 666,n,Playback(/tmp/festival) |
| 02:12.00 | rhollan | exten => 666,n,Return(); |
| 02:15.47 | rhollan | Hmm, it always sees it in extension 's'. Lemme try putting the subroutine there. |
| 02:17.11 | rhollan | Grr. Even that doesn't work |
| 02:27.53 | rhollan | I guess I'll have try M() and code a macro. |
| 02:27.54 | rhollan | Grr. |
| 02:32.18 | kaldemar | you need called-greeting,s,1 |
| 02:32.37 | kaldemar | called-greeting is a context |
| 02:33.19 | kaldemar | s the extension and 1 the priority |
| 02:33.24 | rhollan | So each subroutine has to be the only one in a context? |
| 02:33.59 | kaldemar | no. but they have to be s if U is used. |
| 02:34.23 | rhollan | lemme try |
| 02:36.36 | rhollan | Yup! thanks! |
| 02:37.47 | rhollan | Time for Pizza and Beer |
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| 06:10.12 | DrDigital | who uses vitelity here? |
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| 06:28.34 | schmidts | good morning |
| 06:28.50 | irroot | morning morning |
| 06:33.10 | ChannelZ | DrDigital: I do |
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| 07:06.04 | DrDigital | how does unlimited work? is it unlimited inbound or out? |
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| 07:27.44 | ChannelZ | DrDigital: I don't have it, but the $7.95/mo unlimited is inbound. Don't think they have an unlimited outbound plan |
| 07:29.52 | DrDigital | okay, i was just curious to which way it works, since i pay for a feww |
| 07:30.54 | DrDigital | i just ported my house number over, i dont think it gets very much usage but i had a good number, 941-4101 |
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| 07:32.35 | ChannelZ | yeah I dumped my Qwaste line a year ago and ported to Vitelity. I don't use my home phone very much so I just do the $1.50/mo plan and pay for minutes both ways. |
| 07:32.46 | hetii | Hi:) |
| 07:32.49 | ChannelZ | Which is at least a 10th of what I paid Qwaste |
| 07:33.30 | ChannelZ | hola |
| 07:37.42 | ChannelZ | bed time |
| 07:39.37 | tuxx- | pff no |
| 07:39.39 | tuxx- | just came out of bed |
| 07:39.40 | tuxx- | ;d |
| 07:39.49 | tuxx- | Mon Jun 6 09:39:48 CEST 2011 |
| 07:40.04 | DrDigital | I got this new Laptop and I work from home a lot more so i decided to order cable internet instead of DSL so im ditching the phone company 100% |
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| 08:30.55 | pecenipicek | hello folks. |
| 08:32.00 | pecenipicek | got a quicky question, does anyone know of a SIP softphone that reliably works with TLS and SRTP? Phoner pretends it does, but asterisk just spits this out when i try to make a call from it... "chan_sip.c:8417 process_sdp: We are requesting SRTP, but they responded without it!" |
| 08:33.13 | *** join/#asterisk petn-randall (~lg@p579FB298.dip.t-dialin.net) |
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| 08:38.53 | pecenipicek | noone knows? |
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| 08:54.58 | petn-randall | Hi, ReceiveFAX sometimes bails out with an error. This causes the dialplan to terminate. Is there any way to catch this? |
| 08:56.42 | irroot | what error ?? for fax |
| 08:56.54 | irroot | pecenipicek not used one yet |
| 08:57.08 | pecenipicek | sigh. |
| 08:58.07 | petn-randall | WARNING[18195] app_fax.c: Transmission error |
| 08:58.34 | irroot | fax has certain requirements |
| 08:58.38 | petn-randall | I'm using asterisk 1.6.2.9, so I don't have a debug option with app_fax yet :( |
| 08:59.03 | irroot | 1)only use T.30 on TDM |
| 08:59.12 | irroot | 2)timing is critical for T.30 |
| 08:59.21 | petn-randall | Most of the time it works. Weird enough, even if it throws this error, the fax is still successfully written to disk. |
| 08:59.23 | irroot | use res_fax ?? |
| 08:59.50 | irroot | yeah most the time it will |
| 09:00.42 | petn-randall | However the script which processes the tiff file will never be called if there is an error |
| 09:00.54 | petn-randall | So it has something to do with the flow of the dialplan |
| 09:02.09 | wdoekes2 | petn-randall: I've seen that too.. I Hangup() after receivefax and do processing from the 'h' |
| 09:03.06 | petn-randall | wdoekes2: does asterisk drop to the 'h' extension if an transmission error happens? |
| 09:03.24 | *** join/#asterisk Franckichou (~franck@chy77-1-88-183-41-95.fbx.proxad.net) |
| 09:03.29 | petn-randall | wdoekes2: if so, would you care to share that part of your dialplan? |
| 09:03.34 | irroot | it drops to h when hangup |
| 09:03.46 | irroot | the post processing i always do in h |
| 09:04.11 | irroot | app_fax phase e handler was a little tricky and calls hangup if i remember |
| 09:07.17 | petn-randall | irroot: So using the h extension to process faxes would work around my problem because in both cases the 'h' extension would be called? |
| 09:07.43 | irroot | pretty much |
| 09:09.39 | Franckichou | Hi all |
| 09:10.16 | Franckichou | I've got a question for whoever can answer it : |
| 09:11.18 | Franckichou | Is there something special to set in dahdi config files for asterisk to take te line ? |
| 09:11.51 | Franckichou | bacause I've configured it, I can make a call, but it doesn't takes te calls |
| 09:17.30 | kaldemar | Franckichou: is it an FXO you're talking about? |
| 09:17.47 | *** join/#asterisk Lantizia (~Lantizia@erebus.seaquake.net) |
| 09:17.59 | Franckichou | yes |
| 09:18.51 | kaldemar | Franckichou: put immediate=yes above the channel line, then asterisk will start at context,s,1 immediately when it receives a call. |
| 09:21.32 | Franckichou | kaldemar: Ok I test it |
| 09:25.58 | Lantizia | hey I'm on make menuselect, I'd like to learn about every option in great detail - is each one documented somewhere on asterisk.org? |
| 09:28.58 | *** join/#asterisk engrxyz (~puitpyitr@212.23.51.7) |
| 09:37.29 | pecenipicek | Lantizia, good luck with that :) |
| 09:37.52 | Lantizia | pecenipicek, yeah - the docs look a little... lacking |
| 09:38.02 | pecenipicek | understatement of the year. |
| 09:38.29 | pecenipicek | but basically, you take in the stuff you need, you punt out stuff you dont need and then do the usual lovely make && make install :p |
| 09:38.49 | pecenipicek | actually, you can find what each app and stuff does. |
| 09:39.34 | pecenipicek | check the /doc directory, the AST.pdf should have a listing of everything somewhere in it |
| 09:39.43 | irroot | the safe one is if you know you dont want it dont include it :P |
| 09:40.10 | pecenipicek | well, the Command reference is in there at least. |
| 09:40.20 | pecenipicek | so at least you can approximately glean what you might not need. |
| 09:40.48 | pecenipicek | for anything else that you might have missed, thats what the good old modules.conf and noload => <module name> are for. |
| 09:42.06 | Lantizia | i turn the latest asterisk tar.gz into debian packages - so this'll be used on several systems... kind of want to get all the options exactly as is/might be needed, as future proof as poss |
| 09:42.26 | Lantizia | so curious what they all do (i.e. more than just a four word description of each at the bottom) |
| 09:42.49 | pecenipicek | check the /doc directory. |
| 09:42.51 | pecenipicek | trust me. |
| 09:43.16 | Lantizia | pecenipicek, well ast.pdf doesn't list any of the add-ons lol |
| 09:43.24 | pecenipicek | le sigh. |
| 09:43.33 | pecenipicek | go to the command reference. |
| 09:44.13 | Lantizia | that would be where? |
| 09:44.22 | pecenipicek | its not a nice list of all the things availible in make menuselect, but it describes what each app and function does, so you can cross-reference with stuff in the make menuselect and check what you might not need. |
| 09:45.08 | pecenipicek | page 423, if we have the same versions of it. |
| 09:45.31 | Lantizia | ah i see chan_mobile on that page |
| 09:45.41 | pecenipicek | the pdf should have bookmarks. |
| 09:45.52 | pecenipicek | find the one which says command reference or somesuch |
| 09:46.31 | Lantizia | ok think i have everything i need - will go from here |
| 09:46.32 | Lantizia | thanks |
| 09:46.39 | pecenipicek | no worries. |
| 09:46.49 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-074-249-008.pools.arcor-ip.net) |
| 09:47.16 | pecenipicek | irroot, what do you mean by "F8_ cannot be used,..." here? https://issues.asterisk.org/view.php?id=18674 |
| 09:47.27 | *** part/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 09:50.13 | irroot | er let me peak |
| 09:51.05 | irroot | cant see it on mantis |
| 09:51.14 | pecenipicek | and has the srtp stuff been patched to support the AES_CM_128_HMAC_SHA1_32 cipher or not? |
| 09:51.26 | irroot | ah ok |
| 09:51.39 | irroot | yes can do with the patch |
| 09:51.46 | pecenipicek | i have no problems with pulling stuff from svn's and applying patches manually. |
| 09:52.05 | pecenipicek | (i'd avoid it if possible, but eh, what must be done, must be done :P ) |
| 09:54.01 | pecenipicek | checking out from this link should get me the most bleeding edge asterisk around, amirite? http://svn.asterisk.org/svn/asterisk/trunk |
| 09:55.23 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:55.49 | irroot | so you want the sdp patch ?? |
| 09:55.59 | irroot | the 32/80 taglen patch |
| 09:56.16 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 09:56.27 | pecenipicek | i have no clue to be honest. i know that SRTP doesnt really work with my Yealink T20. |
| 09:56.29 | irroot | but for 1.8 ?? |
| 09:56.42 | pecenipicek | isnt srtp 1.8 only? |
| 09:57.10 | irroot | 1.8/trunk |
| 09:57.27 | irroot | i keep a 1.8 branch of my work too |
| 09:58.43 | pecenipicek | so i check this out then? http://svn.asterisk.org/svn/asterisk/branches/1.8/ |
| 09:59.00 | *** join/#asterisk Cain` (~Geek@unaffiliated/cain) |
| 09:59.32 | irroot | yeah |
| 09:59.50 | pecenipicek | okay. |
| 10:00.05 | irroot | and you can look at /team/irroot/distrotech-customers-1.8 |
| 10:00.33 | pecenipicek | which is? |
| 10:00.41 | irroot | https://reviewboard.asterisk.org/r/1173/ |
| 10:00.48 | pecenipicek | as in, what are the differences from trunk? |
| 10:01.00 | irroot | that is my branch with all the bits i work on and the customers use |
| 10:01.04 | irroot | in the distro |
| 10:01.41 | *** join/#asterisk davlefou (~david@41.225.90.180) |
| 10:01.50 | irroot | the reviewboard link is waht you lookin 4 |
| 10:02.54 | pecenipicek | so, that branch should have the stuff i need? |
| 10:03.14 | *** join/#asterisk Valen (~Valen@ppp121-45-190-216.lns6.syd7.internode.on.net) |
| 10:03.44 | Valen | think its worth getting some g729 licences? small office ~10 staff, lightish phone usage, ~1mbit uplink |
| 10:04.11 | pecenipicek | also, as a side question, you wouldnt have any experience with autoprov file encription, would you irroot? |
| 10:04.13 | Valen | they get some glitches from time to time but QOS would probably help that (but hard to implement in their network) |
| 10:04.14 | WiretapMac | perhaps ILBC would help you |
| 10:05.30 | Valen | G711a G729 G723.1 are the codecs the provider supports |
| 10:05.39 | Valen | currently using G711a |
| 10:05.57 | pecenipicek | i'm all for G711a in general. |
| 10:06.31 | pecenipicek | but it basically boils down to how much bandwith do the calls hog. |
| 10:06.38 | Valen | not enough to bother them |
| 10:06.44 | pecenipicek | we are talking about 10 inbound active calls at one time tops. |
| 10:06.51 | Valen | maximum |
| 10:07.01 | pecenipicek | therefore? |
| 10:07.22 | Valen | i was just wondering if g729 handled the vuagaries of the intertube weather better mainly |
| 10:07.32 | pecenipicek | if it becomes a problem, maybe switch to a codec that uses less bandwith, but its still questionable if its worth the price. |
| 10:07.55 | ectospasm | cost/benefit analysis should be done. |
| 10:08.01 | pecenipicek | codec use wont really change your outbound experience, unless there are heavily congested links involved. |
| 10:08.02 | Valen | its $100 |
| 10:08.11 | Valen | its not that big a deal |
| 10:08.23 | irroot | pecenipicek yip should |
| 10:08.50 | Valen | they have some money they want to spend before end of the financial year so i'm just looking for ways to make things "nicer" |
| 10:09.27 | Valen | I was under the impression that 729 better handled dropped/out of order packets than alaw |
| 10:09.27 | ectospasm | G.711 will have marginally better quality than G.729, but will use eight times the bandwidth. |
| 10:09.39 | ectospasm | that has nothing to do with it |
| 10:09.58 | ectospasm | Valen: no RTP stream handles that "well" |
| 10:10.15 | ectospasm | besides, why would you want to retransmit a packet that ways 400ms ago? |
| 10:10.18 | Valen | no, but there is "better" than "nothin whatsoever" lol |
| 10:10.40 | ectospasm | Valen: I question your understanding of what a codec is |
| 10:10.56 | ectospasm | A codec encodes or decodes from one format to another |
| 10:11.30 | Valen | yes, and some codecs have nifty things like forward error correction and interpolation over missed data |
| 10:11.33 | *** join/#asterisk orn (~orn@rtr1.sh23.sip.is) |
| 10:11.34 | Valen | at least in the wider world |
| 10:11.47 | ectospasm | that's for TCP |
| 10:11.57 | Valen | TCP doesn't do either of those |
| 10:12.27 | ectospasm | no, but TCP has a sense of order of packets, buffering until you get enough of a frame in order, then passing to the upper protocol. |
| 10:12.55 | ectospasm | If a packet in RTP comes in out of order, there's no point in buffering it since the next frame has already been played. |
| 10:12.58 | Valen | forward error correction is when you incorporate extra data into your stream such that a transmission error can be corrected at the receiving end |
| 10:13.13 | ectospasm | Valen: yes, and that doesn't make sense for VoIP |
| 10:13.21 | *** join/#asterisk eerie (hoax@gateway/shell/bshellz.net/x-wjhtsuoxgljrxcsk) |
| 10:13.26 | ectospasm | ...or any Real Time Protocol |
| 10:13.29 | Valen | actually on a lossy link it makes buttloads of sense |
| 10:13.39 | Valen | thats why all the digital TV is done that way |
| 10:13.59 | ectospasm | yeah, if too much buffering occurs, the signal is momentarily lost |
| 10:14.09 | ectospasm | ever see how it gets pixelated? |
| 10:14.26 | ectospasm | that'd be even worse on a voice link |
| 10:14.32 | Valen | that is when the errors in the stream are greater than what the FEC can correct for |
| 10:14.51 | Valen | whats buffering got to do with FEC beyond a packet or 2 |
| 10:17.36 | ectospasm | each frame is 20ms of audio (by default), which is essentially one UDP packet. not much time before the next frame comes in, or time to do any FEC. I've never seen Asterisk described as having FEC, since if the packet is delayed or corrupted enough, you've got deeper issues that an FEC can be expected to correct. |
| 10:18.07 | ectospasm | but then again, I'm not a developer |
| 10:18.14 | Valen | FEC is generally computationally simple |
| 10:18.33 | ectospasm | so? |
| 10:18.41 | Valen | so 20ms is ages |
| 10:19.10 | Valen | all rather beside the point |
| 10:19.38 | Valen | 729 seems to work with 10ms frames btw? |
| 10:19.50 | ectospasm | It can, I think. Lemme look... |
| 10:20.04 | Valen | http://en.wikipedia.org/wiki/G.729 |
| 10:20.36 | ectospasm | just because it's on Wikipedia doesn't make it true, or mean that Asterisk supports all frame sizes... |
| 10:23.13 | *** join/#asterisk davlefou (~david@196.203.154.164) |
| 10:25.09 | Valen | G729 is reputed to perform well in the face of random bit errors |
| 10:27.08 | awk | hi, I have got so used to the way I do dial plans that I have never used Z and N only X.. aka send all calls to X. what is Z and N used for and how would I structure a dial plan with it? |
| 10:27.37 | ectospasm | Valen: you can set the packetization using the syntax (e.g. in sip.conf) with an allow line like this: allow=g729:10 |
| 10:28.03 | ectospasm | Valen: see the doc/rtp-packetization.txt in the Asterisk source to see the step sizes available. |
| 10:28.41 | ectospasm | note that file doesn't seem to exist for Asterisk 1.8 |
| 10:29.38 | ectospasm | awk: Z is everything but zero, N is everything but zero and one |
| 10:30.08 | ectospasm | hence, a typical NANP number: NXXNXXXXXX |
| 10:30.33 | kaldemar | awk: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 10:30.34 | ectospasm | the first digit of an area code or exchange can be [2-9] |
| 10:31.59 | awk | ahh, brilliant, thanks for clearning this up for me.. |
| 10:32.14 | awk | I normally just used [12] etc. |
| 10:37.54 | *** join/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 10:42.20 | *** join/#asterisk rutski (~rutski@96.56.54.186) |
| 10:42.28 | rutski | does anyone else here have a "Digium, Inc. Wildcard TDM800P 8-port analog card", or anything in that family? |
| 10:45.52 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 10:48.56 | *** join/#asterisk carloimperia (~carloimpe@host51-212-static.14-79-b.business.telecomitalia.it) |
| 10:49.21 | pecenipicek | ls |
| 10:49.25 | pecenipicek | bleh, sorry. |
| 10:50.16 | ectospasm | rutski: I don't have one, but I support them (-; |
| 10:50.58 | ectospasm | I mean, I'm pretty sure we have one in the lab... |
| 10:51.38 | *** join/#asterisk carloimperia (~carloimpe@host51-212-static.14-79-b.business.telecomitalia.it) |
| 10:51.52 | rutski | ectospasm: ok, so; perhaps you might be able to help me figure out if I'm right about this basic config issue that I'm having |
| 10:51.59 | rutski | I'm very new to asterisk, to start with |
| 10:52.11 | ectospasm | pastebin your configuration. |
| 10:52.21 | rutski | Last week my boss gave me a new machine in the server closet, and a TDM800P card, and told me to have a go at it |
| 10:52.34 | rutski | ectospasm: here's the system.conf that *works* http://codepad.org/0luvPGhQ |
| 10:52.40 | rutski | here's the one that fails: http://codepad.org/Q2Q0VCTo |
| 10:52.42 | *** join/#asterisk davlefou (~david@41.227.19.24) |
| 10:52.57 | rutski | the problem is that this card is taking input lines from a modem |
| 10:53.06 | rutski | but I think my boss got me the wrong type of card |
| 10:53.17 | rutski | I think this card is built to take input lines only from phones |
| 10:53.25 | rutski | the FXS vs FXO issue I mean |
| 10:53.44 | skrusty | anyone know why, after changing my home router, when i put a call on hold, it doesn't come back? MOH stops for calling party, but the call audio doesn't come back. Worked fine before i changed my router. |
| 10:53.46 | rutski | oops, had an error in the paste: http://codepad.org/2vPMhdgl |
| 10:54.11 | rutski | (forgot to paste the config along with the first link, that 2nd link fixes that; so look only at the 2 most recent links) |
| 10:55.05 | ectospasm | rutski: note that FXO modules use FXS signaling and vice versa, so you'll use fxsks=1 for an FXO port, and fxoks=1 for an FXS port |
| 10:55.18 | rutski | ectospasm: right |
| 10:55.42 | rutski | and when I use fxoks=1 it works |
| 10:55.44 | ectospasm | looks like you've got an FXS port |
| 10:55.49 | rutski | which means I have fxs ports, which is not what I need |
| 10:55.50 | rutski | right |
| 10:56.01 | rutski | the trouble is that all 8 of 'em behave the same way :-p |
| 10:56.15 | ectospasm | then they're all FXS ports |
| 10:56.19 | rutski | so I think he meant to buy an 8-port all-FXO card, but he really got us an 8-port all-FXS card |
| 10:56.25 | rutski | yea, darn |
| 10:56.33 | ectospasm | ...to connect to a POTS line you'll need an FXO module |
| 10:56.37 | rutski | indeed, yeah |
| 10:56.45 | rutski | time to find out if we can return this thing |
| 10:56.49 | ectospasm | actually, the way you sound like you've got it you've got two quad-FXS modules |
| 10:56.56 | rutski | ectospasm: indeed, yeah |
| 10:57.10 | rutski | both of the chips on the card are green colore |
| 10:57.11 | ectospasm | rutski: if it's new, talk to your reseller or distributor, they should be able to hook you up |
| 10:57.12 | rutski | *colored |
| 10:57.19 | rutski | I hear that the FXO ones are red |
| 10:57.28 | rutski | yeah, the trouble is I'm wondering how we got the wrong card to being with, heh |
| 10:57.39 | rutski | my boss tells me that he talked to a sales rep to order the card |
| 10:57.43 | ectospasm | FXO are red, but I always gotta think of it, since I normally have it reversed in my mind (because FXS are powered) |
| 10:57.59 | rutski | ectospasm: indeed, yeah; I know what I mean |
| 10:58.01 | ectospasm | sales rep... where? |
| 10:58.31 | ectospasm | typically if you talk to a Digium sales rep they'll forward you to a local reseller. |
| 10:58.44 | rutski | dunno, good question; have to call him and ask |
| 11:01.59 | *** part/#asterisk awclin (~alinford@g0962184.demon.co.uk) |
| 11:03.00 | skrusty | anyone able to assist? :) |
| 11:03.40 | ectospasm | skrusty: don't ask to ask, just ask. If you don't get an answer within fifteen minutes, don't leave. Hang out for a while, maybe someone will see it when they wake up. |
| 11:03.46 | ectospasm | ~ask |
| 11:03.46 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 11:03.49 | skrusty | i did ask :) |
| 11:04.13 | ectospasm | I was busy assisting others... [looking...] |
| 11:04.25 | skrusty | no worries |
| 11:04.39 | *** join/#asterisk ik_5 (~ik@109.226.17.43) |
| 11:04.43 | ik_5 | hello |
| 11:04.44 | skrusty | im writing the question up to the list too, but thought i would see if any tips came out of here first |
| 11:04.55 | ectospasm | perhaps a NAT keepalive isn't being maintained by the new router << skrusty |
| 11:05.23 | skrusty | it's possible, i will check to see what SIP debug traffic is coming frm the phones |
| 11:05.32 | ik_5 | I'm looking for a way to stream a call/channel in a call to external file while the call is ongoing, does anyone here know if that can be done ? |
| 11:05.38 | skrusty | i have seen this before, it happended on s zyxel router too |
| 11:05.55 | ectospasm | ik_5: MixMonitor() won't work for you? |
| 11:06.18 | ik_5 | ectospasm, afaik the recording saved only when the call is done |
| 11:09.29 | skrusty | ectospasm: when i take the call off-hold, * sends a new SIP OK to the phone, but has to re-transmit it again |
| 11:09.38 | skrusty | trying to figure out the chain of events |
| 11:10.40 | ectospasm | ik_5: what kind of channel is this? |
| 11:11.01 | ik_5 | ectospasm, sip |
| 11:11.02 | ectospasm | skrusty: tcpdump/wireshark to the rescue! |
| 11:12.20 | skrusty | bah :) |
| 11:12.24 | ectospasm | ik_5: yeah, I don't know a way of doing it so you can stream the audio to a file in realtime... |
| 11:12.48 | ik_5 | ectospasm, thanks anyway :) |
| 11:13.12 | ectospasm | ik_5: what kind of application are you trying to build that has that as a requirement? |
| 11:14.24 | ik_5 | ectospasm, some sort of AI to understand the process of a call, the AI people require to listen in "realtime" to the conversation |
| 11:14.31 | ectospasm | ik_5: actually, the Monitor commands shouldn't spool everything in memory before it writes the file, otherwise memory capacity would probably be exhausted quickly for a busy system. |
| 11:14.52 | ectospasm | AI, as in Artificial Intelligence? |
| 11:15.10 | ik_5 | ectospasm, yes |
| 11:15.24 | ectospasm | You could always dump the listener and the other callers into a MeetMe conference |
| 11:15.39 | ectospasm | ...then the listener could do its AI goodness |
| 11:16.13 | *** join/#asterisk davlefou (~david@196.203.147.241) |
| 11:18.11 | ik_5 | ectospasm, i think that monitor just save it somewhere and make it available when done, but i do not know how it works exactly. |
| 11:18.26 | ik_5 | ectospasm, if all else fails then meetme will be the way :( |
| 11:18.45 | ectospasm | ik_5: you'll have to test it out |
| 11:18.49 | ik_5 | but the AI does not know telephony, only audio |
| 11:19.35 | ik_5 | ectospasm, thanks |
| 11:19.39 | ik_5 | i'll |
| 11:19.43 | ectospasm | ik_5: well, the RTP stream coming out of MeetMe will be some audio format, G.711, G.729, GSM, etc. Should be simple to extract it from the RTP stream. |
| 11:20.43 | ik_5 | ectospasm, but i need to identify things per call and specific channel, so I'll have to write something that implement RTP but as a sniffer |
| 11:21.18 | ectospasm | ik_5: sounds like you need to implement your own specialized SIP client. |
| 11:22.26 | ik_5 | i'll look also on other "PBX" such as freeswitch, maybe it can do something to help, before I'll go to that extreme |
| 11:23.59 | ectospasm | at some point you'll need to extract the RTP |
| 11:24.07 | ectospasm | ...with FreeSwitch, or anything else. |
| 11:27.49 | ik_5 | i hope that there is something that already provide that for me |
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| 11:52.09 | *** join/#asterisk hensema (~erik@scrat.hensema.net) |
| 11:52.14 | Lantizia | hey tzafrir_laptop, you know the 1.8.4.2 debian packaging you updated on the svn 3 days ago? if you run './configure; make menuconfig' before you build the package - does it have any effect (i.e. does it actually use the settings you choose when building the debian packages)? |
| 11:52.39 | Lantizia | because I notice it builds the asterisk-mysql even if you have/haven't selected cdr_mysql or app_mysql from the add-ons menu |
| 11:53.58 | tzafrir_laptop | I prefer not to rely on build-time actions, and patch the source instead: |
| 11:54.01 | tzafrir_laptop | http://patch-tracker.debian.org/patch/series/view/asterisk/1:1.8.4.2-1/enable_addons |
| 11:54.05 | *** join/#asterisk Maliuta (~nobusines@kiev.lusan.id.au) |
| 11:54.09 | hensema | hi all, I'm trying to setup a call queue in asterisk 1.8. Everything seems to be working except penalties: only the agent with the lowest penalty is ever called |
| 11:54.23 | hensema | agents with higher penalies never get a call |
| 11:54.31 | irroot | ah hensema that is correct |
| 11:54.49 | tzafrir_laptop | Hmm... that part about format_mp3 is incorrect |
| 11:54.55 | irroot | need to look at /etc/asterisk/orig/queuerules.conf |
| 11:55.46 | tzafrir_laptop | Lantizia, but I just set different defaults. You can use menuselect to override them |
| 11:57.24 | hensema | irroot: erhm, queuerules.conf doesn't seem to do what I need: *always* direct callers to the lowest penelty agents and when those are busy, then direct new calls to another group of agents |
| 11:57.30 | Lantizia | tzafrir_laptop, ok so that patch is already in there so nevermind what is chosen in menuconfig - they're built anyway? |
| 11:57.30 | hensema | irroot: no delay at all |
| 11:57.39 | hensema | it did work in asterisk 1.4 |
| 11:58.10 | tzafrir_laptop | yes |
| 11:58.19 | irroot | hensema perhaps ringinuse setting ?? |
| 11:58.32 | irroot | in queues.conf |
| 11:59.06 | Lantizia | tzafrir_laptop, i told menuconfig to build english core-sounds (all codecs), msuic on hold (all codecs) and extra sounds (all codecs)... but didn't have the resulting .deb files - i'm guessing these .deb's have to be generated seperately? |
| 11:59.29 | irroot | that way when they on a call no additional calls will come in when they busy |
| 11:59.45 | hensema | irroot: I'll try that, thanks |
| 11:59.53 | irroot | that is not 100% certain |
| 12:00.14 | *** join/#asterisk davlefou (~david@41.227.21.57) |
| 12:00.15 | irroot | i have patched 1.8 to have a per agent option and to double check the status ie busy or not |
| 12:01.12 | hensema | doesn't work unfortunately |
| 12:01.24 | irroot | need to reload queues |
| 12:01.32 | hensema | I did |
| 12:01.40 | hensema | 'reload app_queue.so' |
| 12:01.54 | irroot | so when agent 1 is busy it still goes to her with ringinuse = no ?? |
| 12:02.30 | hensema | no, it gets queued |
| 12:02.47 | hensema | so the phone of the busy agent doesn't get another line |
| 12:02.48 | irroot | ah i see |
| 12:03.05 | hensema | it mustn't get queued, it must be connected to an available agent |
| 12:03.15 | irroot | maybe add queuerules to work after short period |
| 12:04.15 | hensema | but queuerules only take effect when the caller enters a queue, so I'll need to get them out of the queue periodically |
| 12:04.40 | hensema | If I need such hacks to make asterisk 1.8 work, I'll just downgrade to 1.4 |
| 12:05.09 | irroot | im pretty sure this was how it worked in 1.4 |
| 12:05.17 | hensema | it didn't |
| 12:06.04 | hensema | in 1.4 I've got AgentCallBackLogin, agent penalties are working |
| 12:06.22 | hensema | in 1.8 we need to dynamically add queue members which is a *pain* |
| 12:06.33 | hensema | callers get stuck in queues with plenty agents available |
| 12:06.45 | hensema | 1.8 plain sucks compared to 1.4 :( |
| 12:08.14 | hensema | in our old setup, when an agent didn't answer within 20 sec, the call would be connected to another agent |
| 12:08.17 | hensema | also broke :( |
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| 12:17.54 | kaii | hensema: sounds like your queue is misconfigured. |
| 12:18.07 | hensema | kaii: I guess so |
| 12:18.35 | kaii | hensema: what strategy and timeout has the queue configured when queried on the CLI? |
| 12:19.28 | hensema | kaii: configured: |
| 12:19.34 | hensema | strategy = ringall |
| 12:19.34 | hensema | timeout = 20 |
| 12:19.34 | hensema | retry = 20 |
| 12:19.34 | hensema | timeoutpriority = conf |
| 12:19.45 | hensema | I don't know how to query on the CLI |
| 12:20.12 | hensema | queue-hxshelpdesk has 0 calls (max unlimited) in 'ringall' strategy (16s holdtime, 577s talktime), W:0, C:84, A:6, SL:0.0% within 0s |
| 12:20.16 | hensema | that? |
| 12:20.21 | kaii | yes |
| 12:20.36 | hensema | (please note I don't understand the timeouts) |
| 12:20.44 | *** join/#asterisk davlefou (~david@41.225.40.18) |
| 12:21.13 | kaii | now, whats the matter with that? you said "callers get stuck in queues with plenty interfaces available" |
| 12:21.26 | kaii | how can that happen, when they all ring at the same time |
| 12:21.27 | hensema | yes |
| 12:21.50 | hensema | no, only the agents with lowest penalty get rung |
| 12:22.10 | kaii | and those are busy and so the caller gets stuck? |
| 12:22.21 | hensema | yes |
| 12:23.15 | kaii | hensema: and after 20 seconds the queue tries again and calls the very same interface again and doesnt move on? |
| 12:24.04 | hensema | kaii: could be, yes (not quite sure how to check this) |
| 12:24.46 | kaii | by looking at the verbose output on the CLI ... you need to reproduce that and look what exactly happens after 20 seconds |
| 12:24.59 | hensema | ok, hold on |
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| 12:27.27 | hensema | ah, yes: the agent is called twice, and after 20 secs asterisk logs "-- Nobody picked up in 20000 ms" |
| 12:28.06 | hensema | so I was wrong before: I said the agent doesn't get called twice, but he does get two lines simultaniously |
| 12:30.09 | hensema | (we tried to avoid this by setting "call-limit=1" in sip.conf, but this breaks call transfers) |
| 12:30.09 | *** join/#asterisk bchia (~Adium@nat/digium/x-ruklyxojehqmhrwq) |
| 12:30.33 | irroot | hensema ringinuse is the neat way of doing it in queues |
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| 12:31.35 | hensema | I |
| 12:31.36 | hensema | I' |
| 12:31.39 | hensema | grrr |
| 12:31.51 | hensema | I'll restart asterisk when all calls are finished |
| 12:32.11 | hensema | maybe the ringinuse setting doesn't get picked up with a reload app_queue.so |
| 12:32.41 | hensema | http://pastebin.com/UxhgXrtj <-- my queues.conf |
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| 12:33.25 | irroot | ringinuse may be per queue |
| 12:33.32 | irroot | let me check the source |
| 12:34.45 | irroot | indeed ringinuse is per queue not genral |
| 12:36.46 | *** join/#asterisk glam (~glam@119.139.184.143) |
| 12:37.13 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:37.13 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:38.40 | irroot | lo there malcolmd |
| 12:38.48 | malcolmd | aloha. another monday |
| 12:40.10 | malcolmd | wrt t.38 relay, in what capacity are your customers using it? you're pushing t.38 to atas at their sites? they're terminating fax from the pstn and handing off t.38 to you? |
| 12:40.16 | irroot | indeed got a tune to brighten it up :P |
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| 12:41.03 | hensema | irroot: ok, tested: even with "ringinuse = no" configured in every queue, the agent is still called twice |
| 12:42.02 | irroot | that is frustrating |
| 12:42.14 | *** join/#asterisk calmh (~jb@irc.nym.se) |
| 12:42.24 | irroot | seen some problems have patched it / worked on it |
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| 12:56.11 | radic | SayDigits() dosn't say the digits |
| 12:56.13 | radic | http://pastebin.com/9ezrgNx1 |
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| 13:07.25 | Chainsaw | leifmadsen: 1.8.4.2 in portage, could you update the channel topic please? :) |
| 13:07.51 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.4.2 (2011/06/04), 1.6.2.18 (2011/04/26), 1.4.41 (2011/04/26), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 13:08.43 | Chainsaw | leifmadsen: Cheers. |
| 13:10.05 | leifmadsen | np |
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| 13:13.46 | puzzled | hi |
| 13:13.46 | *** part/#asterisk calmh (~jb@irc.nym.se) |
| 13:17.25 | Katty | morning |
| 13:18.51 | irroot | hi there katty |
| 13:18.57 | irroot | and puzzlesd |
| 13:19.09 | irroot | liefmadsen top o the morning |
| 13:21.12 | Chainsaw | Good afternoon Katty :) |
| 13:21.16 | puzzled | morning all |
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| 13:45.09 | DelphiWorld | is asterisk still not working with cisco ip phones? |
| 13:45.46 | m4xx | it works for me |
| 13:45.55 | irroot | DelphiWorld perhaps its cisco phones not working with Asterisk :P |
| 13:46.10 | DelphiWorld | irroot: :P |
| 13:46.11 | dr0ck | the bug you refer to is fixed, yes |
| 13:46.23 | DelphiWorld | irroot: maybe you that did breack it up :P :D |
| 13:46.53 | DelphiWorld | dr0ck: in what svn rev? |
| 13:47.02 | irroot | mmm ill blame it on my companion Mr J Daniels |
| 13:47.30 | DelphiWorld | irroot: :D |
| 13:48.36 | DelphiWorld | irroot: PM |
| 13:48.38 | dr0ck | DelphiWorld: i believe it should work in 1.8.4.1, just grab latest tarball 1.8.4.2 |
| 13:48.38 | xxiao | how many calls asterisk can handle on a high-end PC these days? 500? 1000?10000? |
| 13:48.59 | DelphiWorld | dr0ck: i am using svn |
| 13:49.10 | DelphiWorld | xxxmaybe 0 ? :P |
| 13:49.44 | xxiao | read somewhere saying 250 calls, this weekend hope i have time to try kamailio+asterisk |
| 13:49.53 | dr0ck | DelphiWorld: then just svn up to latest should be fine |
| 13:50.07 | DelphiWorld | X-Rob: let me get a cisco phone first :P |
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| 13:50.38 | DelphiWorld | hi putnopvut |
| 13:50.56 | putnopvut | DelphiWorld: howdy |
| 13:51.20 | DelphiWorld | putnopvut: how is going, mark? |
| 13:51.26 | putnopvut | Going all right |
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| 14:12.40 | pigpen | I saw the previosu post about kamailio+asterisk. I have been deploying asterisk for some time and I am a bit in the dark about the need to integrate kamailio. |
| 14:13.00 | pigpen | Is it kamailio is a load workhorse with sip, and asterisk is the brains? |
| 14:13.11 | *** join/#asterisk hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e) |
| 14:14.18 | irroot | basically |
| 14:14.34 | irroot | kamalio does nice load balancing and fail over |
| 14:14.47 | irroot | does the maintaining of sip registrations |
| 14:15.37 | pigpen | k. so it helps keep the sip maintenance off the asterisk load....which can be a good thing on high volume. |
| 14:16.02 | KavanS | anyone have any word on the skype/asterisk sitch? - why it was killed etc? |
| 14:16.05 | KavanS | pro tips, drop em. |
| 14:16.35 | florz | pro tip? don't use proprietary technologies |
| 14:16.45 | irroot | kamalio talks bout thousands of users registering on single box |
| 14:17.03 | pigpen | irroot, k. |
| 14:17.06 | irroot | only active calls will be seen by asterisk |
| 14:17.10 | KavanS | florz, hardy har har - I bet you've got something proprietary in your stack Mr. Stallman |
| 14:17.11 | pigpen | KavanS, google it. |
| 14:17.18 | KavanS | florz, don't go watching DRM dvd's :) |
| 14:17.30 | KavanS | googling... |
| 14:17.39 | KavanS | figure since this was #asterisk, someone might have something more than the internet has |
| 14:17.40 | KavanS | oh well. |
| 14:18.01 | florz | KavanS: no, of course I don't buy DRM!? |
| 14:18.18 | pigpen | irroot, I like the idea. I can see this would be a very good thing for public sip accounts, but I have not needed anything yet and have a couple of deployments of asterisk sitting about 500 sip devices. |
| 14:18.22 | pigpen | each. |
| 14:19.18 | irroot | yeah our ITSP [we are supplying soon too] use it all over have for ages |
| 14:20.16 | pigpen | good info to know, thanks for the quickie. |
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| 14:44.38 | drmessano | KavanS, because Skype wants everyone to use the SIP door now |
| 14:44.52 | KavanS | oh really? |
| 14:44.53 | KavanS | ok |
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| 14:49.47 | sxpert | drmessano: one can use the reverse engineered skype door ;) |
| 14:50.10 | drmessano | Their SIP offering has $$$$ all over it. You pay monthly for the channel itself, then for your subscriptions.. They could also care less about Skype to Skype connections. Seems to be all about Skype becoming another SIP based ITSP |
| 14:50.28 | Kobaz | oh they have sip now |
| 14:51.10 | drmessano | sxpert, which is pretty much useless. It's a reverse engineer of a much older version of the protocol, and has huge gaps. Won't communicate with others using newer client versions, etc |
| 14:51.19 | sxpert | ah. crap ;( |
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| 14:58.03 | drmessano | Kobaz, IMO, Skype for Asterisk does TOO much, and doesn't give Skype a monthly piece of pie for every channel like they have in place for SIP. I think they want those looking to blast their network with PSTN calls to use the SIP connectivity so they can play ITSP, and for those wanting to build apps to access the network, they have a nice little limited API as part of their SDK.. which falls short of "build your own client"... |
| 14:59.34 | Kobaz | yeah |
| 15:00.32 | drmessano | Besides, even though the pre-Microsoft Skype was the one that killed it, Microsoft would have done so |
| 15:00.40 | drmessano | It was only a matter of time |
| 15:02.40 | Katty | hi danny |
| 15:02.47 | drmessano | Even though Skype took years to open up the little bit of the network they finally have to developers, it looked like they had finally started to warm up to the idea and found a revenue stream for it. Microsoft won't have any interest in pushing that further |
| 15:02.54 | drmessano | Hi Katty :) |
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| 15:08.25 | Katty | what's the word |
| 15:09.15 | beek | Hi Katty |
| 15:09.27 | Kobaz | bird? |
| 15:09.34 | dr0ck | pecksniffian |
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| 15:09.40 | Kobaz | bird is the word |
| 15:11.36 | Katty | hugs beek |
| 15:14.24 | beek | smiles |
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| 15:19.53 | leifmadsen | Kobaz: bird bird bird, bird is the word, b- b- b- b- bird bird bird bird, bird is the word |
| 15:20.08 | Kobaz | everybody knows! |
| 15:21.07 | Kobaz | leifmadsen: i love this hangup patch i found |
| 15:21.18 | Kobaz | i think it was from oej |
| 15:21.19 | leifmadsen | Kobaz: which one? |
| 15:21.37 | Kobaz | adds one log statement on ast_queue_hangup and one log statement in ast_softhangup |
| 15:21.46 | Kobaz | i never thought of doing that, now i know which channel hangs up first |
| 15:23.00 | leifmadsen | nice |
| 15:23.13 | leifmadsen | mark it as tested and that it works so we can put it into review and maybe get it in for 1.10 |
| 15:23.29 | Kobaz | k |
| 15:23.53 | Kobaz | i know that's something people are been asking for |
| 15:24.01 | Kobaz | to know which channel hung up first |
| 15:24.07 | Kobaz | could probably add some code to write it to the cdr |
| 15:25.09 | Kobaz | and it also sends a manager event for each spot too |
| 15:25.17 | leifmadsen | Kobaz: yep indeed, could be useful for recordings and such too |
| 15:25.35 | leifmadsen | people have asked me that for queues as well to make sure the agents aren't hanging up the call :) |
| 15:26.03 | Kobaz | ah |
| 15:26.03 | Kobaz | yeah |
| 15:26.16 | Kobaz | and i'm still getting: |
| 15:26.17 | Kobaz | [2011-06-06 11:04:49] ERROR[16168]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x0 for 0xb5f07d38 |
| 15:26.20 | Kobaz | [2011-06-06 11:04:49] ERROR[16168]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x0 for 0xb5f07d38 |
| 15:26.23 | Kobaz | that sort of stuff... from musiconhold |
| 15:26.40 | Kobaz | killing me, i still can't officially move to 1.8 |
| 15:26.51 | Kobaz | that problem exists in 1.6.2 also |
| 15:28.49 | Kobaz | actually, and I think i've seen that in late 1.6.0 as well |
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| 16:43.55 | thehar | 34 |
| 16:45.14 | Qwell | /lose |
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| 16:46.16 | thehar | seriously |
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| 18:38.11 | pdtpatrick | Question for you smart folks -- I get this message all day |
| 18:38.12 | pdtpatrick | chan_dahdi.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 |
| 18:38.26 | pdtpatrick | i've rebooted the box.. no go |
| 18:38.42 | pdtpatrick | has anyone gotten that problem and how did they fix it? Could be a bad PRI ? |
| 18:39.11 | nate_h | hey anyone know a good place to sell some VOIP hardware? I have some new in box/and used cisco conference phones, and a few other new voip phones. |
| 18:39.24 | Qwell | nate_h: ebay? |
| 18:39.49 | nate_h | Qwell, ya i just figured maybe there was a community page somewhere that more VOIP people would see vs ebay |
| 18:46.48 | WIMPy | pdtpatrick: Bad line/connection, IRQ processing troubles or anything else that causes errors or overruns. |
| 18:47.47 | pdtpatrick | can u please suggest some troubleshooting techniques if possible? i've done most of what i could think of |
| 18:48.20 | *** join/#asterisk TJ_stamm (~TJ_stamm@66.254.218.126) |
| 18:49.09 | WIMPy | That's a little hard as it could be anything. Does the card share an IRQ? |
| 18:49.28 | pdtpatrick | cehcking |
| 18:49.30 | pdtpatrick | checking* |
| 18:51.11 | cusco | hey folks |
| 18:51.54 | cusco | is it possible to dial a PRI number like 210000000#000 where 210000000 is the DID and 000 will be a variable in asterisk? |
| 18:52.15 | Qwell | cusco: some lines have support for things like that |
| 18:52.17 | WIMPy | Sure |
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| 18:54.30 | cusco | well I'm using dumpchan() and I don't see any 000 in it |
| 18:54.31 | cusco | :( |
| 18:55.00 | WIMPy | Did you add XXX to your extension? |
| 18:55.22 | cusco | ow... I need that? |
| 18:55.32 | cusco | ok let me try |
| 18:55.55 | WIMPy | You need to make that part of the extension and then cut the digits from EXTEN. |
| 18:57.39 | bchia | cusco: There is a free video outlining dialplan variables on this page that describes the process of selecting substrings from a variable http://www.digium.com/en/training/courses/#essentials |
| 18:57.47 | cusco | Accepting call from '910898179' to '210332456' on channel 0/15, span 5 |
| 18:57.58 | cusco | so its not adding the extra digits to exten |
| 18:57.59 | cusco | :( |
| 18:58.09 | cusco | probably telco doens't support it |
| 18:58.53 | WIMPy | Im'n not sure, but you might have to enabe overlap receice in chan_dahdi.conf. |
| 18:59.05 | cusco | googles it |
| 18:59.14 | WIMPy | Too hot to type :-( |
| 18:59.49 | *** part/#asterisk clintc (~clintc@n128-227-97-92.xlate.ufl.edu) |
| 19:00.17 | WIMPy | Hmm. Looking at it, it mught always do it on receive. |
| 19:01.34 | WIMPy | You can look at the pri debug. If the setup contains a Sending Complete, you telco probably doesn't allow you to receive further digits. |
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| 19:03.24 | SuperNull | Hey All, Need info on Asterisk 1.2 dialplan .. specifically if its possible to add Alert-Info .. or add custom headers to sip messages. we are kind of locked into this software due to some physical gear connected to it so upgrading is not a solution unfortunately |
| 19:03.29 | cusco | pri debug shows: |
| 19:03.30 | cusco | Called Number (len=12) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '210332456' ] |
| 19:03.35 | cusco | http://paste.debian.net/119036/ |
| 19:04.38 | WIMPy | There it is. |
| 19:05.20 | cusco | only the 9 digits |
| 19:05.33 | cusco | not the '#111' |
| 19:05.36 | cusco | let me try * |
| 19:06.02 | WIMPy | You are unlikely to be able to receive * or #. |
| 19:06.09 | WIMPy | Stick to digits. |
| 19:06.26 | WIMPy | Anything else is almost certainly filtered. |
| 19:06.36 | cusco | well Im pretty sure that dialing just 210332456111 won't work |
| 19:06.53 | WIMPy | Either that or no go. |
| 19:07.01 | cusco | it did! |
| 19:07.40 | cusco | but no info on the 111 |
| 19:07.45 | cusco | http://paste.debian.net/119039/ |
| 19:08.19 | cusco | DNIDDigits= 210332456 |
| 19:08.34 | WIMPy | Bad luck then. Maybe you can gat that as an option. |
| 19:09.01 | cusco | well I will have to use ivr instead |
| 19:09.02 | cusco | lol |
| 19:09.12 | cusco | I just added overlapdial=yes to chan_dahdi.conf |
| 19:09.13 | pdtpatrick | WIMPy -- no it is not sharing IRQ |
| 19:09.17 | cusco | but only reloaded config |
| 19:09.21 | WIMPy | Ot talk to your telco. |
| 19:09.26 | cusco | seems that I need to unload the module and load it again |
| 19:09.37 | cusco | that will take dahdi calls down... |
| 19:09.45 | cusco | I could try and ask telco yes |
| 19:09.47 | p3nguin | I can't imagine there's anything in Asterisk 1.2 that would keep you locked in to it. |
| 19:10.08 | p3nguin | Anything Asterisk 1.2 had would have been improved in later branches. |
| 19:10.17 | WIMPy | pdtpatrick: Is there a chance the system might miss IRQs for other reasons? IDE disks? Single core? |
| 19:11.42 | pdtpatrick | i don't think so .. don't see anything after checking anyway. 8 cores btw |
| 19:12.43 | WIMPy | That's quite a lot. |
| 19:13.59 | pdtpatrick | right |
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| 19:24.53 | cusco | I called telco, they say I must talk to our comercial agent |
| 19:24.54 | cusco | gah |
| 19:25.04 | cusco | sent e-mail... lets wait |
| 19:25.47 | cusco | it would be nice for a dial gateway, to dial ${DID}${DST} and bridge the call to ${DST} |
| 19:25.57 | cusco | we woudln't need boring ivr for that |
| 19:25.58 | cusco | :P |
| 19:26.36 | WIMPy | Depending on where you're calling from, you might be able to use a subaddress for that as well. |
| 19:26.53 | cusco | subaddress ? |
| 19:27.33 | WIMPy | An optional parameter to any number. |
| 19:28.07 | WIMPy | But you need to be able to dial that, off course. |
| 19:30.17 | cusco | so you mean using overlap |
| 19:30.41 | WIMPy | That's not related. |
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| 19:31.00 | cusco | I don't understand |
| 19:31.08 | cusco | using only 9 digits as did |
| 19:31.26 | cusco | ah.. like the last digit, from 0 to 9 where 0 is dst0 and 9 is dst9 ? |
| 19:31.51 | pdtpatrick | anyone know why i keep getting this error? |
| 19:31.51 | pdtpatrick | http://pastebin.com/y6n7LQ5v |
| 19:32.05 | WIMPy | No, you can have an additional parameter. But you need to be able to dial that in the first place. |
| 19:32.55 | WIMPy | pdtpatrick: No need to use a PB for one line. But you might want to get your line quality checked. |
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| 19:33.51 | pdtpatrick | something i could do myself or need to have someone come out ? |
| 19:34.18 | ariel_ | hello everyone |
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| 19:35.08 | ariel_ | tzafrir_laptop: quick Xorcom question, on the Astribank 4 port PRI/T1/E1 unit can you mix the E1 and T1 on same unit? |
| 19:35.23 | tzafrir_laptop | yes |
| 19:35.36 | ariel_ | tzafrir_laptop: t/y |
| 19:35.50 | tzafrir_laptop | see the sample /etc/dahdi/xpp.cong |
| 19:35.58 | tzafrir_laptop | xpp.conf |
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| 19:36.24 | WIMPy | pdtpatrick: Depends on the NT you've got. Some can do BERTs. |
| 19:36.26 | ariel_ | will do |
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| 19:55.38 | WIMPy | What do I have to enable to get iax netstats? |
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| 19:59.41 | beetlejooz | Anyone here work with digium cards? |
| 20:00.24 | WIMPy | ~ask |
| 20:00.24 | infobot | Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
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| 20:01.37 | beetlejooz | How do I know which pair of wires off my 66 block connect to which channels of a digium card. It's a 25 pair amphenol cable connecting the block to the card. |
| 20:01.52 | beetlejooz | I've already scoured the manual |
| 20:01.58 | beetlejooz | AEX2400 |
| 20:02.09 | beetlejooz | It doesn't discuss wiring |
| 20:03.54 | beetlejooz | As much as I would love to ditch the digium cards and go pure IP, this is what I have to work with. Any help is appreciated. |
| 20:06.22 | beetlejooz | wow...like 100 peeps in here and nobody knows? ;) |
| 20:07.05 | beek | beetlejooz: http://www.telephonecentral.com/custom.aspx?id=28 |
| 20:12.26 | bobb_WU | beetlejooz, is your question already answered? |
| 20:12.35 | beetlejooz | not really |
| 20:12.48 | beek | beetlejooz: Did you look at that site? |
| 20:12.58 | bobb_WU | i work with the digium cards at work |
| 20:13.11 | bobb_WU | and that 25 pair color guide that beek posted is exactly what you're looking for |
| 20:13.47 | beetlejooz | How do you know where to terminate to the 66 block though? That's where I get confused |
| 20:13.54 | bobb_WU | ah |
| 20:14.03 | beetlejooz | I have analog phones that come in to 66 block and then we cross connect them over |
| 20:14.10 | beetlejooz | I inherited this setup |
| 20:14.15 | bobb_WU | 66 blocks, wow we don't use those any more |
| 20:14.24 | bobb_WU | we're on the 110 block and the 24 pairs pretty much fills it up |
| 20:14.25 | beek | beetlejooz: Start at the top and work your way down. |
| 20:14.43 | bobb_WU | so using the color guide, just map out the first one and work up to the 24th one- |
| 20:14.43 | pdtpatrick | QUestion -- what does "b8zs mean in this line for dahdi ? span=1,2,0,esf,b8zs |
| 20:14.50 | beetlejooz | cross connecting from one 66 block to the other |
| 20:15.00 | beetlejooz | out of the 2nd into the digium |
| 20:15.06 | beek | beetlejooz: Forget about your phones for a second. |
| 20:15.09 | beetlejooz | ok |
| 20:15.14 | WIMPy | pdtpatrick: It's the line code. |
| 20:15.39 | bobb_WU | punch the cable down to its own block, then do your crossconnects from there |
| 20:15.46 | beek | beetlejooz: Wire your 66 block following that guide with the wires just as listed. You'll have an empty pair at the bottom. |
| 20:16.10 | beek | Now cross-connect whatever pair you want to whatever phone you want. |
| 20:16.20 | bobb_WU | exatly |
| 20:16.25 | bobb_WU | *exactly |
| 20:18.05 | beetlejooz | It's a prewired block with an amphenol port on it |
| 20:18.23 | beetlejooz | it's already in use |
| 20:18.27 | beetlejooz | I'm trying to add more stations |
| 20:18.47 | *** join/#asterisk mountainm2k (~msturtz@gw.booyahnetworks.com) |
| 20:19.05 | beek | beetlejooz: One side of that thing already has 24 pairs connected, doesn't it? |
| 20:19.27 | beetlejooz | correct |
| 20:19.41 | beek | Alright then. Use that guide to identify pins. |
| 20:20.18 | beek | Hook your phone to the other side of the 66 block connecting to the correct color wires for whatever pair (channel) you want them connected to. |
| 20:20.28 | beek | Then use bridging clips to finish the job. |
| 20:20.39 | beek | Are the phones connected to the other side? |
| 20:20.51 | beetlejooz | they come into a different block |
| 20:21.03 | bobb_WU | that's good |
| 20:21.22 | beek | Okay. Then use cross-connect wire to extend the connection from one 66 block to the other. |
| 20:21.23 | beetlejooz | phone company lines also come into a different block |
| 20:21.38 | bobb_WU | do you know which ones on the other blocks are supposed to correspond to the dahdi channels? |
| 20:21.46 | beek | ARRRRGH! |
| 20:21.50 | beetlejooz | lol |
| 20:21.59 | beetlejooz | bobb: that's my original question :P |
| 20:22.06 | bobb_WU | oh gotcha |
| 20:22.13 | beek | That's simple: |
| 20:22.16 | bobb_WU | well that's a matter of choice |
| 20:22.43 | bobb_WU | are the phones labled in the closet? ( or do you have tone generator/wand? ) |
| 20:22.44 | beek | Look at the color code I gave you. Figure out what pair that is. That corresponds to the DAHDI/xx channel. |
| 20:23.04 | beetlejooz | I know where the phones come in on their corresponding block, that's not the problem. |
| 20:23.13 | beek | Once you know that, trace the wire from the TDM card block to the phone block. |
| 20:23.18 | beetlejooz | or rather, that's not my hangup (no pun there0 |
| 20:23.33 | bobb_WU | beek: that raises a side question for me- can you make calls to dahdi/01? or is it required to be dahdi/1 ? |
| 20:23.55 | beek | I never tried two digits, only one. |
| 20:24.03 | bobb_WU | beetlejooz: have you created a dialplan yet? that's going to be your mapper |
| 20:24.19 | beetlejooz | There's an existing one |
| 20:24.39 | beek | Here's a backward way to do it. |
| 20:24.57 | beetlejooz | I just added some SIP connections so that should save me some grief figuring out how to cross connect to the phone company |
| 20:24.58 | bobb_WU | an asterisk dialplan? |
| 20:25.01 | beek | Turn on Asterisk logging at the console. Pick up a phone, make a call, see what channel it's coming from. |
| 20:25.15 | beetlejooz | beek: I did that, I know the existing channels |
| 20:25.25 | beetlejooz | for the phones in use |
| 20:25.28 | beek | Then you know which ones are unused. |
| 20:25.59 | X-Rob | bobb_WU, the stuff after the slash is just a number, not a string. So /0000000001 and /1 are exactly the same. |
| 20:26.04 | beetlejooz | Yes, just needed a way to figure out where to punch the phone down on the 66 so that I get to the right channel on the digium |
| 20:26.17 | beek | Are your channels 1-24? |
| 20:26.23 | bobb_WU | thanks rob. i wasn't sure and the way it outputs from our db just annoys me :) |
| 20:26.25 | beetlejooz | yes |
| 20:26.53 | beek | Then if you have an available channel at, say, channel 6, look at the color chart on that website that I sent to you. |
| 20:27.17 | *** join/#asterisk Ryushin (proxy@cl-412.phx-01.us.sixxs.net) |
| 20:27.27 | beek | You'll see that pair 6 corresponds to the "red/blue" and "blue/red" pair. |
| 20:27.36 | beek | That's where you'd hook up one end of your cross connect. |
| 20:27.58 | beek | Either that, or count pairs until you get to the sixth pair. |
| 20:28.34 | beek | Starting from the top of the block, or the end that is not missing a pair. |
| 20:29.21 | beetlejooz | well looking at the digium manual it uses like pin 1&26 for port 1. Is that already worked out in the cable? |
| 20:29.29 | beek | Yes. |
| 20:29.37 | beek | Telco stuff is standardized. |
| 20:29.55 | beek | Pin 1 is on the one side of the connector, Pin 26 is directly opposite. |
| 20:30.04 | beek | Ignore that.... just go by the color code and you'll be fine. |
| 20:30.05 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 20:30.15 | beetlejooz | awesome...TYVM! |
| 20:30.46 | beek | Just be sure that the pairs are wired as shown in the color chart. If they aren't, track down the installer and kill him/her. |
| 20:31.17 | beek | You'll find them to be okay, I'm sure. |
| 20:31.42 | beek | Even if they aren't, just go by the color code. Digium did it right. |
| 20:32.07 | beetlejooz | sweet...ty |
| 20:32.13 | beetlejooz | afk a min |
| 20:33.38 | bobb_WU | haha how true that is beek. death to bad installers! |
| 20:33.49 | otwieracz | What happend, If I have nat=yes while not being behind NAT? |
| 20:34.01 | bobb_WU | as long as you don't kill people for messy wiring, cause then i'm in trouble |
| 20:35.53 | bobb_WU | its quittin' time. cya eveybody! |
| 20:37.44 | p3nguin | otwieracz: Sometimes nothing bad. |
| 20:40.00 | otwieracz | Hmm⦠|
| 20:40.08 | otwieracz | I have 2 clients. |
| 20:40.14 | otwieracz | Everyone can do echo-test. |
| 20:40.25 | otwieracz | But noone can hear second one. |
| 20:41.21 | mountainm2k | Question: Is there an easy way to authenticate a VM subscriber in a way that will fail if the extension and password match? I have a door-phone, and I have a seperate method by which asterisk can run a shell command which buzzes the door, but I'd like to give employees the option of buzzing themselves in IF they know their voicemail password... |
| 20:42.20 | mountainm2k | Seems like I probably need to run a shell script to do this for me, because VMAuthenticate just authenticates, it won't care if they haven't changed their password |
| 20:55.03 | WiretapWork | bobb_WU, I kill people for messy wiring |
| 20:55.50 | WiretapWork | mountainm2k, match callerID against their vm password before allowing them entry? |
| 20:58.16 | mountainm2k | WiretapWork: callerID will always be the door phone -- I would put this feature only in the door phone context |
| 20:58.58 | mountainm2k | But since the door is effectively outside (elevator lobby, which is unsecured) -- I want people to enter their voicemail user and pass |
| 20:59.15 | mountainm2k | but since half the company doesn't bother to set up their voicemail -- pretty big security flaw |
| 21:00.06 | mountainm2k | I can probably just collect user and pass, and hand those out to a shell script which will look in the database... |
| 21:00.58 | WiretapWork | or you could authenticate with some other credentials |
| 21:01.07 | WiretapWork | perhaps implement 2FA :P |
| 21:01.15 | mountainm2k | could, but it'd need to be accessible to * |
| 21:02.19 | mountainm2k | since people can't be bothered to bring their damn access card with them, I dunno how likely they would be to bring a 2FA token with them |
| 21:02.26 | mountainm2k | assuming we had those, which we don't -- LOL |
| 21:02.31 | WiretapWork | lol, fair call |
| 21:02.53 | mountainm2k | if people would bring their damn badge with them when they go to the can after hours, none of this would be neded |
| 21:03.11 | WiretapWork | I thought you were replacing access cards |
| 21:03.11 | mountainm2k | no way jose |
| 21:03.11 | WiretapWork | but now this makes more sense |
| 21:03.11 | WiretapWork | you're dealing with morons :P |
| 21:03.24 | WiretapWork | just use a biometric door phone? they can be had reasonably cheap |
| 21:03.25 | mountainm2k | aren't we all??? |
| 21:03.37 | mountainm2k | cheap != free |
| 21:03.40 | WiretapWork | people won't forget to bring their fingers with them I'd hope |
| 21:03.42 | mountainm2k | plus requires enrollment |
| 21:04.40 | mountainm2k | I'm going for easy -- remember, morons |
| 21:04.55 | WiretapWork | lol, true |
| 21:05.25 | WiretapWork | the craziest thing I ever saw |
| 21:05.38 | WiretapWork | was a callcentre with bulletproof glass blocking off everything but the reception area |
| 21:05.48 | WiretapWork | and a pin+fingerprint+card entry scanner |
| 21:06.06 | mountainm2k | "Kill the receptionist, save the monkey!" |
| 21:06.14 | WiretapWork | on the 5th floor of a building that already required keycard entry |
| 21:06.42 | mountainm2k | and somebody could just tazer the receptionist and hop over the desk I'm guessing? |
| 21:06.53 | WiretapWork | bearing in mind that armed robberies happen about 5 times per year in the entire country XD |
| 21:07.12 | mountainm2k | what country r u in? hahah |
| 21:07.26 | mountainm2k | also I guess depends on the call center |
| 21:07.29 | WiretapWork | mountainm2k, nope, receptionist was completely seperated off, glass wall behind her too |
| 21:07.35 | WiretapWork | and it was a tech support callcentre |
| 21:07.37 | WiretapWork | I'm in NZ |
| 21:07.46 | mountainm2k | yeah, for tech support that's just dumb |
| 21:07.57 | mountainm2k | we have a receptionist, she sits in the first cubicle |
| 21:07.59 | WiretapWork | I was there for a job interview |
| 21:08.04 | mountainm2k | front doors are all unlocked durring the day |
| 21:08.21 | WiretapWork | a guy in business attire interviewed by 5 guys in polos and jeans |
| 21:08.24 | mountainm2k | if you look like you belong, or if you go the other direction from her, you're in |
| 21:08.34 | WiretapWork | I later learned they told my recruitment agent I was too casual |
| 21:09.23 | WiretapWork | between that and the next week's police-interrogation style interview by a certain large scale GPS chip manufacturer, I've never seen so many oddities XD |
| 21:09.26 | mountainm2k | had that discussion with my new boss when he first started |
| 21:09.33 | mountainm2k | he believes "casual" is relative |
| 21:09.58 | mountainm2k | polo and jeans OK, as long as its tucked in, no holes, hair combed, shaved as appropriate, etc... |
| 21:10.12 | mountainm2k | somebody walks in here in a suit, I automatically don't trust 'em |
| 21:10.13 | mountainm2k | lol |
| 21:10.34 | mountainm2k | Guess we're off-topic from Asterisk though |
| 21:11.36 | WiretapWork | lol |
| 21:11.41 | WiretapWork | I'm with you on this one |
| 21:12.12 | WIMPy | What do I have to enable to get iax netstats between two Asterisks? |
| 21:13.18 | pigpen | netstats? |
| 21:13.59 | pigpen | iax2 show netstats |
| 21:14.01 | pigpen | ^^ |
| 21:14.07 | WIMPy | iax2 show netstats |
| 21:14.13 | pigpen | beat you. |
| 21:14.18 | pigpen | ;-) |
| 21:14.28 | pigpen | I forget about that command |
| 21:14.36 | WIMPy | Yes, but I don't get any values there on calls between two Asterisks. |
| 21:14.45 | pigpen | hmm... |
| 21:15.14 | WIMPy | On a call from iaxcomm I get information. |
| 21:15.15 | pigpen | I think you have to have jb enab.ed. |
| 21:15.43 | pigpen | but I have been wrong one other time this month, so it is possible I could be wrong again. |
| 21:16.05 | WIMPy | Oh, you do it by month? :-) |
| 21:16.39 | pigpen | My wife would be laughing her ass off right about now..... |
| 21:16.53 | WIMPy | Ok, so as it's forwarded to SIP, I guess that meand I need to forcejitterbuffer? |
| 21:18.13 | pigpen | http://www.voip-info.org/wiki/view/Asterisk+cli+iax2+show+netstats |
| 21:18.29 | pigpen | ^^^^ "stats may not be complete if the remote end isn't using the new jitterbuffer" |
| 21:19.24 | WIMPy | Yes, forcejitterbuffer did it. |
| 21:20.24 | pigpen | ah, cool. |
| 21:21.17 | WIMPy | Well, not that cool. The jitterbuffer is pretty senseless there. |
| 21:24.43 | *** join/#asterisk d_preston215 (~chatzilla@static-76-161-250-54.t1.cavtel.net) |
| 21:26.14 | *** join/#asterisk kclever (~kclever@68-185-2-158.static.mdfd.or.charter.com) |
| 21:29.11 | kclever | Ok, I have a noob question. I'm allowing an outside Polycom dial in to our internal Polycom. Audio works great, but I'm not getting any video. |
| 21:29.44 | kclever | Using SIP of course.. And Device to Device is working when registered to the server. |
| 21:30.09 | kclever | Anyone have somewhere they could point to? |
| 21:30.28 | pigpen | I take it you have opened up various ports through your fw? |
| 21:31.06 | kclever | Yeah.. I completely opened both sides of the network to see if that was the issue |
| 21:31.50 | pigpen | so: polycom ---NAT---------NAT-------polycom? |
| 21:32.27 | pigpen | if so, you may want to setup a site to site vpn. |
| 21:32.29 | kclever | No.. Polycom <--> Asterisk <--> Polycom |
| 21:32.40 | pigpen | So all a local lan/wan? |
| 21:32.43 | pigpen | no internet? |
| 21:32.47 | pigpen | no firewalls? |
| 21:33.07 | pigpen | you said: "I completely opened both sides of the network" |
| 21:33.38 | kclever | Polycom <--- {Internet} --> Asterisk <--{internal Network} --> Polycom |
| 21:34.21 | pigpen | so Polycom "A" has a public address? |
| 21:35.20 | kclever | I'm using URI Dialing for the Outside Polycom.. Yeah I totally opened up iptables and turned off SELinux stuff.. |
| 21:36.25 | kclever | I don't believe it does (not mine, unfortunately) but we have been successful connecting directly in the past. |
| 21:37.02 | pigpen | Ok, so more than likely it has some sort of firewall on the remote side. |
| 21:37.16 | pigpen | You may want to reboot the router/firewall/modem, whatever. |
| 21:37.35 | pigpen | Also, it is possible an isp involved may be blocking some traffic. |
| 21:37.36 | WIMPy | Hmm. Do you have videosupport enabled? |
| 21:37.38 | pigpen | just some ideas. |
| 21:38.04 | pigpen | kclever, so did the video work before as well? |
| 21:48.53 | *** join/#asterisk luckman212 (~irc@2001:470:1f07:1225:5ce6:4f07:90b5:8bbe) |
| 21:53.47 | jdoe | why might asterisk instantly cancel a call after receiving a 183? |
| 21:54.43 | *** join/#asterisk Defraz (~Defraz@corp.fuzecore.com) |
| 21:59.00 | *** join/#asterisk Syre_x (~syrex@dsl-146-17-172.telkomadsl.co.za) |
| 22:01.51 | *** join/#asterisk jhmichelle352 (~jhmichell@41.69.251.53) |
| 22:02.02 | jhmichelle352 | kobaz: ping |
| 22:02.07 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 22:02.33 | jdoe | ... hrm. And a side-effect of making mantis read-only is that you can't search it now. Applying a filter = error. |
| 22:03.05 | jhmichelle352 | sorry wrong screen |
| 22:07.10 | *** join/#asterisk WiretapWork (~Wiretap@unaffiliated/wiretap) |
| 22:12.49 | *** join/#asterisk fskrotzki (~fskrotzki@cpe-67-253-245-174.rochester.res.rr.com) |
| 22:14.39 | jdoe | the version is 1.8.3. As part of the call flow I get a 183 from my provider, ~6 seconds of nothing, then "Scheduling destruction [...] in 6400ms", followed by an *immediate* CANCEL. |
| 22:15.10 | jdoe | is that, er, expected behaviour? Is the log message just delayed? and is there a way I can increase that timeout? |
| 22:38.39 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
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| 23:25.09 | SunTsu | yay, asterisk 1.6.2.18 finally is in netbsd's pkgsrc and yay again, signalling is fixed, so all sip phones are told when one picks up a call. |
| 23:26.58 | WiretapWork | you mean hinting |
| 23:27.25 | SunTsu | maybe, yes |
| 23:38.47 | *** join/#asterisk ariel_ (~chatzilla@pdpc/supporter/active/abatista) |
| 23:41.54 | el3slave | dont necessarily need a layer 3 switch for voip traffic, corrrect? |
| 23:42.14 | el3slave | as long as its managable with QoS |
| 23:44.48 | *** join/#asterisk pushpop (~pushpop@pool-173-77-230-33.nycmny.fios.verizon.net) |
| 23:44.52 | *** join/#asterisk adolfomaltez (~taro@190.86.145.63) |
| 23:58.37 | showme | no reason to use a layer3 switch, some people will use a switch that supports 802.1q (vlans), or as you said a switch that supports QoS, but it all really depends on the number of phones, and the other activity on the LAN. Where most people hurt for doing voip is really their internet speed. |