IRC log for #asterisk on 20100908

00:00.57*** join/#asterisk Mhaddog_Mac (~anonymous@adsl-32-170-232.mia.bellsouth.net)
00:08.28*** join/#asterisk cpw11 (~cpw@CPE0002b3c126e6-CM0011ae0163ee.cpe.net.cable.rogers.com)
00:09.08rift0rneone?
00:09.14cpw11hello?
00:11.05WIMPyrift0r: That's rather unspecific. Maybe you should read the Book.
00:11.06WIMPy~book
00:11.07infobot[~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
00:13.11cpw11hi, is this the right place to ask a question about dahdi?
00:14.03WIMPy~ask
00:14.03infobotQuestions 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.
00:16.29cpw11heh ok. i'm trying to debug why my new AEX800 just refuses to work for more than about a day at a time. it dies because of the "Missed interrupt. Increasing latency to x ms in order to compensate." at least, it accumulates a few of those until x=25, then we get "ERROR: Unable to service card within 25 ms and unable to further increase latency." and finally, and this seems to be the sign to reload the kernel module: "Hos
00:16.51rift0rWIMPy i dont have a zapata device, and I have my trunk defined in sip.conf and specified in the extensions.conf to dial to SIP/mycontext
00:17.04rift0rbut for some reason it tries to go through a zap interface
00:19.05cpw11more: i have removed the xen vm that this was originally running in (with pci passthrough), but in fact, running dahdi in dom0 seems to have made the problem significantly worse, not better (i got about 2-4 weeks of uptime vs 1-2 days since)
00:19.10WIMPycpw11: Lost interrups are often due to interrupt sharing or the system being too slow.
00:19.31rift0rI have TRUNK=SIP/bandwidth.com_outbound specified in extensions.conf and exten => _XXXXXXXXXXX,1,Dial(${TRUNK}/+${EXTEN}) specified in my outgoing context
00:20.29cpw11the system is a quad core xeon- it's about 2 years old, with about 6gb of memory. the other zapata card (pci and 5 years old) has run fine in it, without a single complaint. it's the new aex card i got to replace a borked t1/channelbank that's being the problem
00:22.01cpw11there is interrupt sharing going on, but i notice that the zap card isn't using msi interrupts, so all interrupts are going to cpu1, which isn't very efficient
00:22.58cpw11i would like to try and force either distribution or moving to a non-shared interrupt, but the older tricks i know don't seem to work on the new motherboard.
00:25.40cpw11the reading i've done indicates that you can't change the interrupt of a pciexpress board, because pciexpress is supposed to use msi interrupts, not old style ones
00:25.50cpw11but clearly, that's not true for this aex800 board
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00:40.49rift0rok got it working
00:41.07rift0rapparently it didnt like $trunk and when I manually specified the full context it worked
00:41.08rift0r=+
00:41.17rift0r6dd
00:41.27rift0rodd
00:41.40rift0rAnyways, now to fix my rtp issue and one way audio
00:41.41rift0r:(
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00:57.43ChannelZoh, bother.
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02:01.16mifadirHello
02:01.40mifadirwhat's the best softphone with higth resolution ????
02:01.42mifadirfor video
02:08.56*** join/#asterisk hackeron (~hackeron@gentoo/user/hackeron)
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02:10.42hackeronhey, I'm trying to set up asterisk with festival - I have festival --server running, I start firefox, dial an extension and hear nothing :( - I see "client(5) Wed Sep  8 03:03:43 2010 : accepted from localhost" in the festival log and -- Executing [500@default_local:1] Festival("SIP/reception1-00000003", "Test to see if this festival thing works") in new stack -- not seeing any errors anywhere and text2wave works fine - any ideas?
02:10.57hackeronI start asterisk I mean, not firefox - it's late :)
02:11.37mmatticeanybody know of a relatively cheap door phone that works with *?
02:12.27p3nguinhttp://www.door-entry.co.uk/
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02:21.53*** join/#asterisk kalimc (~mcurry@proxy.hostopia.com)
02:25.23kalimcHere is my chan_dahdi.conf, for some reason we're unable to dial out - but can receive calls on channel 1.  http://pastebin.com/sD3EiivY
02:25.50kalimcanything I can do to debug?
02:31.42WIMPyMaybe you should tell us what happens?
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02:33.43p3nguinIT DOESN'T WORK!  What more could you possibly need to know to fix it?
02:35.16WIMPyWhere I can access the machine and where to send the bill :-)
02:38.03kalimcIf  you are serious, we could pay
02:38.55kalimcWe need to make two PRI channels into 1 group.  I can give you a number to call to speak to my manager, and we will pay.
02:39.27WIMPyIf I have to travel to you, the's going to be a massive bill. So if I where you, I'd rather provide more information.
02:39.57p3nguinWhy travel?  He's on the public intertubes.
02:40.08WIMPyMaybe.
02:40.30WIMPyBut a description would be a good idea.
02:41.40zxvffno one is around in the #freepbx channel, maybe soeone here wants to take a look at this for me? I added a new PRI card to my PBX (new = additional) and now i'm getting "all circuits busy" error for incoming or outgoing calls
02:41.41kalimcOk, so, we are able to get in and out bound calls with 1 PRI
02:41.41zxvffhttp://pastebin.ca/1935474
02:41.44zxvffthis is what i'm getting
02:41.44WIMPyIf one direction works, it can't be that bad.
02:42.19kalimcAs soon as we activate and configure the second PRI in dahdi, we get this- "[Sep  7 22:40:16] NOTICE[7589]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (1) Hangup"
02:42.32WIMPyzxvff: Probably the new card is found first so all your channels have been renumbered.
02:42.57kalimcOur current dahdi conf is here: http://pastebin.com/sD3EiivY
02:43.43kalimcwe are dialing DAHDI/G1/xxx
02:44.08hackeronhey, I have verbose = 10 and debug = 10 in [options] in asterisk.conf but I still don't see verbose messages in asterisk -R -- any ideas?
02:44.13WIMPykalimc: Do you have a backup D channel on the 2nd interface?
02:44.21zxvffWIMPy: i don't think tha'ts the case. i had that problem initially, one of the other tech guys went in and fucked with the config and rebooted the system, and now none of it works
02:44.29zxvffcan i check which channels are showing up where?
02:44.38zxvffalso in asterisk CLI i have no dahdi* commands
02:44.40kalimcWIMPy: We have one on 24, and 48
02:45.48WIMPyzxvff: Well, then it obviousely isn't loading dahdi any more. You can try to increase debug and verbose and do a module load chan_dahdi.
02:46.25WIMPykalimc: Two active D channels or one primary and one backup? The the two interfaces share the same numbers?
02:46.56kalimcThe 2 lines share the numbers, I can ask about the backup
02:46.59WIMPyzxvff: Do you have more dahdi hardware in that machine?
02:47.58kalimcWIMPy: We have a primary dchan and a backup
02:48.04WIMPykalimc: Sounds ok to me then, but I guess T1 can have more strange setups than E1.
02:48.27kalimcdoes my config look ok?
02:48.30kalimchttp://pastebin.com/sD3EiivY
02:49.04zxvffWIMPy: yes, 2 pri cards and an 8 port analog card
02:49.06zxvffhowever it works
02:49.09zxvffi'm not sure how i fixed it
02:49.14zxvffeverything just started working
02:49.15zxvffweird
02:49.26zxvffi did module load dahdi, and it told me it couldn't load the module
02:49.29zxvffand then it worked
02:50.05WIMPyzxvff: That dos not sound too convincing.
02:50.13zxvffi know
02:50.14zxvff:/
02:50.16zxvffi don't know what happened.
02:50.25kalimchow should I go about solving the problem with the dchans?
02:52.40hackeronhow do I debug AGI? -- I tried to add verbose and debug to logger.conf, but all I really see is: <SIP/reception1-00000000>AGI Script tts.agi completed, returning 0
02:52.44WIMPykalimc: So it stops working when you _configure_ the 2nd i/f?
02:52.51hackeronbut the script fails to run :(
02:53.44kalimcWIMPy: yes, we can receive calls on G1, but sending out we get [Sep  7 22:40:16] NOTICE[7589]: pbx_spool.c:339 attempt_thread: Call failed to go through, reason (1) Hangup
02:54.37WIMPyBut dialling out is ok, with only one i/f?
02:54.58kalimcyes
02:55.14kalimcif we pull the second cable everything is good
02:55.23kalimcbut only have 23 channels
02:55.47WIMPyAh, so it's also working with both i/f configured, but one unplugged?
02:56.11WIMPyI guess it's time for pri intensive debug.
02:56.59kalimcsweet
02:57.11kalimchow do I enable that
02:57.20hackeronanyone?
02:58.37WIMPyIt differs a little between versions. pri set intensive debug span 1. If in doubt use the tab key.
02:58.49zxvffhmmm so only one dahdi span is configured properly i guess
02:59.04zxvffhow do i add an additional dahdi span to my asterisk conf?
03:00.03kalimcok coo
03:00.56WIMPyzxvff: Probably via your GUI.
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03:03.01asteriskmonkeyanyone test t38 with asterisk and a metaswitch before?
03:03.29zxvffnot sure where
03:04.29zxvffoh, edit /etc/asterisk/chan_dahdi_custom
03:04.48WIMPyI guess that's why I try to avoid things that alledgly make things easier.
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03:07.45unspinwith asterisk 1.6.2, what the function to cause asterisk to pause for a specified amount of time before continuing with the next item?
03:08.13asteriskmonkeyive got digium paid fax licenses and im having a tad issue with a provider.. wonder if someone wouldnt mind helping me check my configs im getting errors like refused to negotiate T.38 and Audio FAX not allowed on channel
03:08.21unspinit used to be wait, but that isn't working for me
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03:17.05Guggeunspin: core show application Wait
03:17.14Guggewhat does that output?
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03:38.06asteriskmonkeyanyone on that knows t38 well ?
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03:50.33coppiceif you've paid, then ask digium's support
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05:48.47schmidtsgood morning all
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07:22.44xiacloHello? this thing working?
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07:23.59schmidtsdepends on what you expect to work?
07:24.26xiacloirc in general, never been on this before
07:25.05xiaclohaving an issue getting TLS working on asterisk, can't seem to find anything on google to fix it
07:27.03xiacloI keep getting "tcptls.c:218 handle_tcptls_connection: FILE * open failed!" ... googling just keeps bringing me back to the source code, which I don't really understand
07:27.35xiacloanyone have any ideas?
07:27.40schmidtssorry i have never used tls so i dont know anything about this ;)
07:28.28xiaclothanks anyways :)
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07:31.30zplinuxhi all
07:31.54zplinuxcan I transcode voicemail messeges to mp3 before sending them to the email?
07:32.06zplinuxif not, where  may I suggest it?
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07:34.23schmidtsyou could but you have to do it by your own
07:34.49schmidtsi am not sure whats the name of it, but you can launch an external script to send the voicemail, in this script you can transcode it to mp3
07:36.03xiacloif you can call a custom script (no idea) you can easily toss it through ffmpeg like so "ffmpeg -i voicemail.wav -o voicemail.mp3" assuming ffmpeg is compiled with lame support
07:41.48schmidts;externnotify=/usr/bin/myapp is what you want
07:42.18schmidtshave a look at the configs/voicemail.conf.sample file in your asterisk directory there its descipted how to use this
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08:33.36OlafsenMquestion: AGI get data returns even if dtmf is received just before audio prompt is started
08:43.24ChannelZThat was a question?
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08:45.34OlafsenMguys
08:45.36OlafsenMsor?
08:45.38OlafsenMso?
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09:11.25timahvo1Hi am trying to configure an E1 line on an Digium, Inc. Wildcard TE121 single-span T1/E1/J1 card (PCI-Express) (rev 11) but dahdi_scan shows it as type=digital-T1
09:11.53ectospasmtimahvo1: you probably need to set the T1/E1 jumpers appropriately.
09:11.53timahvo1anyone know how I can select type E1 because as I understand it this should be selectable ?
09:12.08timahvo1ectospasm: how is that ?
09:12.12ectospasmtimahvo1: or, you can use the t1e1override module parameter
09:12.56ectospasmtimahvo1: if the jumpers are present, bridging the pins, it's in E1 mode.  If the jumpers are missing, or in storage mode (only on one pin), it's T1
09:13.13ectospasm(see the TE120 series manual for details)
09:13.31timahvo1ectospasm: thanks
09:13.54ectospasmno problem
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09:33.43schmrzHello Everyone! Is there an cli program that I can use on Windows to connect to Asterisk ?
09:33.57schmrzThank you :-)
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09:42.50ectospasmschmrz: putty.exe
09:43.29ectospasmgoogle putty download
09:43.30upbms comic chat aswell
09:43.38upb# APPEARS AS ANNA
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09:58.12datacompboyHi everybory! How to receive HEX numbers over PRI? From telco i'm receive [70 10 a1 32 31 31 37 33 36 33 33 30 35 00 30 30 30 36] == where code "00" is instead HEX number "B". So phone cuts off on that zero, and i can't route based on full number :(
09:58.27datacompboyis that problem in *, or on telco side?
10:00.26ectospasmdatacompboy: are these DTMF digits, or raw pieces of the HDLC frame?
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10:01.03datacompboyectospasm: phone number in "Message type: SETUP" frame
10:01.29ectospasmdatacompboy: is that the called number or calling number?
10:02.09ectospasmpri debug will tell you both if they're both in the SETUP
10:02.43datacompboyectospasm: that called number
10:03.27datacompboyectospasm: http://pastebin.ca/1935638
10:04.18ectospasmwhat does from-pstn look like?
10:04.57datacompboyectospasm: exten => 2117363305B0006,1,...
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10:05.24ectospasmdatacompboy: that's wrong
10:05.41ectospasmdatacompboy: it's trying to match 2117363305 as the extension
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10:06.12datacompboyectospasm: yep, i see that. and problem is because "B" in number is come there as "\x00"
10:06.15datacompboyand asterisk cut it off
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10:06.54kamhhi all
10:06.55kamh:)
10:07.02timahvo1ectospasm: ok the t1e1override works when I reload the driver modules by hand, is there a way I can specify the override options in /etc/dahdi/modules ?
10:07.25ectospasmtimahvo1: try an options line in /etc/modprobe.d/dahdi.conf
10:08.02ectospasmdatacompboy: I still don't see where the B is coming across, it's not in the SETUP
10:09.06datacompboyectospasm:  [70 10 a1 ] = header, [ 32 31 31 37 33 36 33 33 30 35 00 30 30 30 36]  = phone. 2 1 1 7 3 6 3 3 0 5 ?? 0 0 0 6
10:09.45datacompboyectospasm: on place of byte 00 was "B" in source call
10:10.28tamieldatacompboy: your pastebin is what you are receiving from telco ?
10:10.40ectospasmI'm not sure, seems like the sending system shouldn't be doing that
10:10.53datacompboytamiel: yes
10:13.46tamieldatacompboy: so, called number is truncated (at 'B') by telco
10:14.37datacompboytamiel: not. it come full, but instead of "B" there was zero byte
10:14.50datacompboyand asterisk cut off phone number at zero byte
10:16.31tamieldatacompboy: ok so probably telco equipment is replacing letters by 00 ...
10:16.36timahvo1ectospasm: thanks man , looking good now :)
10:17.58tamieldatacompboy: look in libpri q931.c in functions receive_called_party_number and q931_get_number
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10:19.48datacompboytamiel: yep. it read it as string with length, and use as zero-terminated string
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10:25.12*** join/#asterisk tgreer (~tso@gaius.tsone.info)
10:25.33tgreerhey all... Looking for recommendation for a ISDN card suitable for asterisk
10:26.21tamieldatacompboy: making a dirty hack (replacing 00 by 'B') in q931_get_number ... ? :(
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10:54.49ectospasmtimahvo1: no problem!
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10:55.17ectospasmtgreer: how many spans (ports) do you need?
10:55.56tgreerectospasm: 1 tbh
10:56.10ectospasmtgreer: do you have a PCI/PCIe preference?
10:56.19tgreerectospasm: PCI... iirc the box doesnt have PCIe
10:56.38ectospasmtgreer: Digium offers the TE122 with optional echo cancellation
10:56.47ectospasmtgreer: disclaimer: I work for Digium
10:57.28ectospasmhttps://www.digium.com/en/supportcenter/documentation/viewdocs/TE122 << tgreer
10:57.50tgreerectospasm: do i need echo cancellation on an ISDN line?
10:57.55ectospasmtgreer: http://www.digium.com/en/products/digital/te122.php
10:58.05darkskiezthats a PRI card, hes looking for a BRI card :)
10:58.21tgreersorry i shoulda been more specific
10:58.27ectospasmdarkskiez: ISDN is too ambiguous
10:58.35ectospasmI assumed PRI
10:58.42darkskiezectospasm: yeh, but i was chatting to tgreer on another chan :)
10:58.53ectospasmdarkskiez: d'oh (-;
10:59.05tgreerwell hold on..
10:59.07ectospasmtgreer: there's the B410P, or the HA8
10:59.12tgreeri'd assume its a BRI card i need?
10:59.43ectospasmtgreer: do you need BRI (2B + 1D channels) or PRI (23/30B + 1D channels)
10:59.56tgreerectospasm: i dont know? I need to get a line fitted...
11:00.00darkskiezwoah, that HA8 is a beast, nice.
11:00.01Chainsawtgreer: "ISDN2e" -> BRI
11:00.06Chainsawtgreer: "ISDN30" -> PRI
11:00.10tgreerah ok
11:00.15tgreerso it'll be BRI
11:00.45Chainsawtgreer: You say PCI. How old is this machine?
11:00.54tgreerChainsaw: not that old...
11:01.00tgreerit'll be getting upgraded in future
11:01.06tgreerbut i kinda need to prove it works first
11:01.12tgreerbefore boss will spend out more on it
11:01.14Chainsawtgreer: Please be specific, I am trying to find out if you have a PCI 2.2 capable mainboard.
11:01.22tgreerChainsaw: 2 secs
11:01.26Chainsawtgreer: Because if you do not, you are entering a whole world of pain.
11:02.03darkskiez^-- not an understatement
11:02.49tgreerpulling mobo model from hwinfo atm
11:03.07Chainsawtgreer: Okay.
11:04.16tgreer<PROTECTED>
11:04.17tgreersigh
11:04.21tgreerlet me go open the case
11:04.25ChainsawNo need.
11:04.32ChainsawPastebin me the full output of dmidecode please.
11:04.36ChainsawI can generally figure it out from there.
11:04.37hrhrhrha8 seems like overkill
11:04.49hrhrhrmay as well go pri if you've got that many circuits
11:05.06darkskiezunless of course you needed to connect to internal equiptment
11:05.37ectospasmhrhrhr: the jump from BRI to PRI in some jurisdictions may be prohibitively expensive
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11:06.12ectospasmespecially if the telco has to change the switch at the CO to accomodate PRI
11:09.43darkskiezits about £100/mo for 8 lines on a PRI in the uk
11:10.18Chainsawdarkskiez: I believe my 3 BRIs were slightly cheaper then that.
11:10.18ectospasmdarkskiez: 8 lines, eight physical PRI links, or 8 PRI channels?
11:10.23tgreerscratch that
11:10.28tgreerit has pci express too
11:10.32tgreerhttp://www.asrock.com/mb/overview.asp?model=g31m-s
11:10.35darkskiez8 pri channels
11:10.39darkskiezon 1 pri
11:10.47Chainsawtgreer: Consumer-grade kit really...
11:11.03tgreerChainsaw: yes
11:11.06Chainsawtgreer: But yes, that'll work fine with either PCI or PCI-express cards. Just pick one that you have a free slot for.
11:11.12tgreerboth
11:11.33darkskiezi was scared today coz last night a lightning strike took out power to my asterisk server, and it handnt been rebooted in 2.5 yrs
11:11.35ectospasmtgreer: I'd go with an HB8 with B400M, so if you move to a production system later you can use the PCIe card
11:11.41Chainsawtgreer: Then I'd pick PCIe so you have PCI slots free for any specialty hardware.
11:11.49darkskiezbut it came back no problems =]
11:13.24ectospasmdarkskiez: lucky you!  Is the system on UPS?
11:13.59tgreerectospasm: uh yeh...
11:13.59darkskiezyes, on two phases
11:14.03tgreerbit outside my budget
11:14.09darkskiezbut power went out for hours.
11:16.44ectospasmtgreer: where are you located?
11:17.05ectospasmtgreer: you may be able to find a local reseller/distributor that can provide a substantially lower price.
11:17.10tgreerectospasm: UK
11:18.01ectospasmtgreer: Telappliant is the UK distributor, they may be able to hook you up with an HB8 at a significant discount from the Digium web store price
11:18.37ectospasmif you contact Digium sales they may be able to connect you with a local reseller, too
11:19.03ectospasm...but yeah, these cards aren't exactly cheap
11:19.20tgreeryeh i noticed. heh
11:19.31tgreeri can get one at £75
11:19.36tgreerbut not a digium one
11:20.44ectospasmI'm curious as to who makes it...
11:21.00tgreeravm
11:21.45tgreerhttp://www.avides.co.uk/AVM-FRITZ-Card-v2-0-20001708/dp/B00029PNAQ?traffic_src=froogle&utm_medium=organic&utm_source=froogle
11:22.36ectospasmHopefully it doesn't go on the fritz
11:22.47tgreerwas waiting for that..
11:22.49ectospasm...and with that, you only get one span of BRI
11:23.00tgreerthats all i need?
11:23.18ectospasmall Digium BRI adapters come with four spans, plus optional echo cancellation
11:23.30ectospasmtgreer: it depends on what your application is
11:23.40tgreerectospasm: 1 line into our PBX
11:23.49tgreerchoice of analog or isdn
11:24.20ectospasmthen a single-span card is all you need
11:24.38tgreeras i said earlier :)
11:24.48ectospasm...and ISDN will give you two voice channels
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11:26.45ectospasmtgreer: do you know if that Fritz card is DAHDI-compatible?
11:27.11tgreerectospasm: its recommended in http://www.asteriskguru.com/tutorials/bri.html
11:28.01tgreers/recommended/known to work/
11:28.24tgreerthats fucking sweet... I've not seen a bot do that before
11:28.41ChainsawIt's an infobot thing.
11:28.49tgreerlol
11:28.56ectospasmtgreer: Oh, it uses mISDN, that's why
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11:29.25ectospasmGetting mISDN to work with Asterisk 1.6.2.x is nontrivial IIRC
11:30.17tgreersooo what you're saying ectospasm is i can a) pay more for a card and drop is straight in... or pay less for the card and slave my ass off getting it to work
11:30.50ectospasmtgreer: essentially, there's always a tradeoff
11:31.02ectospasmtgreer: who knows?  It may just work
11:31.17tgreerectospasm: i was looking to spend £200 on a card
11:31.31darkskiezhow much was the sangoma you saw
11:31.54tgreer310+VAT
11:32.07tgreerthat includes a 2 port bri channel card
11:32.19ectospasmtgreer: also, how much support are you expecting to need?
11:32.29tgreerectospasm: wrt?
11:32.46ectospasmtgreer: installing and configuring the card, getting it working with Asterisk
11:32.58tgreerectospasm: i'm not a total mong...
11:33.03tgreerso ur
11:33.06tgreer*shrugs*
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11:34.04tgreerfirst time i've ever implemented a asterisk server with a physical line
11:34.10ectospasmtgreer: I can't vouch for other suppliers' support capacities
11:34.48tgreerectospasm: and i cant afford to spend £550 on a card
11:35.16ectospasmtgreer: understood.  What is this system's primary function going to be?
11:35.19tgreerif digium done a single slan pci/pcie BRI card i'd have it
11:35.42tgreerectospasm: main server... PBX/File Server/rsync server...
11:36.02tgreerto support 3-10 users eventually
11:36.06ectospasmtgreer: so it will be in production
11:36.14tgreerectospasm: yes
11:36.20tgreerwell will be once i buy kit heh
11:37.09ectospasmhow tolerant of PBX downtime is your organization?
11:37.28tgreerectospasm: not tollerant...
11:37.48tgreerconsidering it'll be our main lines
11:38.17ectospasmtgreer: I'm not saying choose Digium, but be very certain you can obtain live support for whatever BRI hardware you install
11:38.33tgreerunderstandable..
11:38.45ectospasmyou may want to have a spare card on hand in case it fails.
11:38.48tgreerjust cant actually justify the cost of their cards heh
11:38.51tgreerectospasm: intend on...
11:39.18ectospasmah, so £550 is really £1100 (-;
11:39.19oelewapperkeis there something like AGI, but for SIP SUBSCRIBE requests ?
11:39.44tgreerectospasm: thats just for 1 pbx
11:39.54tgreeri'm setting up a 2nd one in the phillipenes within 6 months
11:40.08tgreeralthough i doubt they have ISDN there
11:40.08tgreerlol
11:40.13Chainsawtgreer: With the kit they're pricing up here... I would just go for a separate gateway: http://www.mapleleaf-technologies.co.uk/products/Patton_VoIP_Routers_BRI.php
11:40.15ectospasmtgreer: you'd be surprised
11:40.21oelewapperkesomething that gives end-users an address to SUBSCRIBE to, but that the actual online-offline is controlled from within an application
11:40.32Chainsawtgreer: In which case it will talk SIP to your Asterisk box and all your slots remain free.
11:41.13ectospasmChainsaw: I bet there's a million different ways to do it.
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11:41.52tgreeryeh i'm just trying to keep everything in the 1 box
11:42.00Chainsawtgreer: Separate gateways have better fallback options.
11:42.12tgreerChainsaw: yeh... try explaining that to the boss...
11:42.17tgreertheres 3 people in the company
11:42.25tgreeri dont think he's gonna spend millions on it
11:42.37Chainsawtgreer: Just give the 600 pound option that the others gave and the ~300 pound box that I suggested.
11:43.10tgreerectospasm: how much do i have to pay for the live support from digum? £25/minute?
11:43.32ectospasmtgreer: with one of our hardware devices, it's free for five years (-;
11:43.44ectospasmtgreer: it's covered under install support
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11:43.50tgreerfuck me you may have just sold it
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11:44.37ectospasmtgreer: the one thing install support doesn't cover is dialplan assistance.  We will demonstrate basic call functionality, but the rest is up to you (-;
11:44.55hrhrhr[12:31] <ectospasm> tgreer: who knows?  It may just work  <---- tehehe
11:45.12hrhrhri found the b410p a steep learning curve
11:45.18hrhrhrbut once working, it was flawless
11:45.52topriddy_could someone please recommend a GSM voice/data modem i could use with javax.comm / AT commands... I want a medium where i can auto dial AT commands from computer server
11:46.05tgreertopriddy_: ubinetics GC201
11:46.09ectospasmhrhrhr: try the HA8/HB8.  I *still* haven't mastered the curve, and I have had training on it!
11:46.49tgreertopriddy_: with a pcmcia <> pci bridge like the Elan P111... I have it set up as a text message server
11:47.14tgreerectospasm: tbh the way the boss is going... the dial plan is gonna take weeks to write lol
11:47.26topriddysomething I can plug in a normal SIM card module...so i can do stuff like check balance and auto load recharge cards
11:47.44tgreertopriddy: the GC201 is just a standard sim card slot
11:48.01tgreerhttp://www.amazon.co.uk/UbiNetics-GC201-PCMCIA-Mobile-Phone/dp/B00005NT8J
11:48.14tgreerand no the earphones are not a requirement for it toow ork :P
11:48.45ectospasmtgreer: we do offer open source subscriptions that cover dialplan, but only the highest level provides consultative services
11:48.49topriddytgreer: and it should fit my purpose...i.e make calls and send sms...earphones can be plugged to computer...basically i just want something i can send various STANDARD* AT Commans too
11:49.19tgreertopriddy: yeh thats what i use it for
11:49.31tgreeruse AT commands to send/receive SMS
11:49.41tgreeri use some FOSS called smstools to do it
11:50.40tgreerectospasm: i'm sure i'll be fine for dial plans
11:51.09ectospasmtgreer: just letting you know we do offer the service (-;
11:51.22tgreerectospasm: thanks... will bear that in mind if i mong out..
11:51.35topriddytgreer: okay....I have looked at smslib its FOSS too, but basically I would need to do stuff manually myself through AT commands...thanks again. Whats the smstools link???
11:51.59tgreerhttp://smstools.meinemullemaus.de/
11:52.26tgreerectospasm: so let me confirm... i buy a B410P card... I can get it for £550...
11:52.35tgreerthat gives me 5 years live support?
11:52.39OlafsenMwaaaarriiiooooorsssssss
11:52.41tgreeris it 24/7/365 or what
11:52.47OlafsenMcome out to playeeeeyyeey
11:52.55OlafsenM:)
11:54.11ectospasmtgreer: you get business hours support, that's your business hours.
11:54.27tgreerhmmm
11:54.35tgreerfree for 5 years you say?
11:54.48ectospasmtgreer: at the current time we have technicians staffing the phones from 7pm Sunday through 7pm Saturday, US CDT (UTC-5)
11:54.57ectospasmtgreer: yes, free for the warranty period of the card
11:55.43tgreerectospasm: fantastic... is this documented anywhere on the digium website to show to the boss?
11:55.45ectospasmtgreer: note that the hours aren't 24/7, and they are subject to change.  If you call our EMEA office right now, and choose support, you'll get me
11:55.53ectospasmtgreer: lemme find a link
11:56.03topriddytgreer: thanks for the links
11:56.07tgreertopriddy: np
11:56.24dacm_workHi guys,
11:56.53tgreertopriddy: if you decide to get it and u get the pcmcia brige i suggested (elan p111) give me a shout and i can help u set it up... I'm normally on #scotlug
11:58.35dacm_workThis is slightly off topic but... I'm intending to build an IP-PBX system for our office. Some of our guys go out to smoke and I'd like them to be able to take calls outside. It's also important that we can easily transfer calls between each other and put people on hold. Can anyone here suggest any reasonably priced phones that would allow this?
11:59.47tgreerdacm_work: normal DECT (cordless) phone with ATA bridge
12:00.08tgreeror something like snom m3
12:00.10tgreerhttp://www.snom.com/en/products/voip-dect-phones/snom-m3-voip-phone/
12:00.23tgreerif that covers the area required
12:00.38dacm_workThere area isn't very big.
12:00.46dacm_worktgreer: Thanks.
12:01.14dacm_workIs it possible to get ones that will display caller info (from XML?) as well?
12:01.23tgreerdacm_work: snom <3 XML
12:01.32dacm_workCool :-)
12:01.33tgreerbut double check the datasheets on the m3 or m9 cordless
12:01.38tgreernot dealt with them before
12:01.55dacm_workOk.
12:03.16tgreerectospasm: the sites not that confusing is it? :P
12:04.21ectospasmtgreer: nah, I was just trying to find it stated explicitly.  I can't find it, except in our warranty policy:  http://www.digium.com/en/company/view-policy.php?id=Warranty-Policy
12:04.27tgreerah ok
12:04.38ectospasmtgreer: look under Hardware Warranty
12:05.17ectospasmthe warranty is five years, and it's implicit that you can call us up to five years after the purchase.
12:06.14tgreerectospasm: i dont suppose you could send me an official email stating it?
12:06.22ectospasmIf you need me to write up some text explaining this, I can do so.  Unfortunately I can't create a support ticket without having a registered card.
12:06.31ectospasmtgreer: pm your address (-;
12:07.06tgreerpm'd thankies
12:09.38dacm_worktgreer: With these DECT IP phones, if I want to run an asterisk IP-PBX and control which handset rings, can I get away with buying additional handsets or do I need to buy many bases units?
12:10.09tgreerdacm_work: i'm not 100% sure
12:10.17tgreeras i said, i've never dealt with the cordless SNOMs before
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12:10.29dacm_worktgreer: Oh ok.
12:10.49dacm_worktgreer: Thanks for all the help anyway.
12:11.00tgreerdacm_work: no probs... sorry i cant be more help
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12:13.37ectospasmdacm_work: the way the DECT handsets I've seen work, the base station for the wireless handset only services one handset, but I suppose you could have one that services multiple handsets
12:16.13tgreerectospasm: i was just looking at the snom website
12:16.32tgreer9 handsets per base... 4 simultanius calls per base
12:17.18ectospasmtgreer: ah, I don't know that much about VoIP handsets.  I can handle Polycom, but I would need to dig for anything else (-;
12:17.59tgreerdacm_work: yeh looks like it just acts as an ethernet <> DECT bridge...
12:18.42dacm_workectospasm: Well I'd noticed that they often sell additional handsets. I don't know if that is just to replace broken ones, or if both handsets always ring at the same time (lots of non IP DECT phones work like this), or if the handsets could be controlled individually.
12:19.04tgreerdacm_work: only way to check would be call them?
12:19.29dacm_worktgreer: Yeah I guess. I was having a look through their support forums.
12:19.46tgreerfrom as much as i can work out from the datasheet
12:19.50tgreerthe base is just a bridge
12:20.02tgreerand each individual handset can be configured differently
12:20.33dacm_worktgreer: Well if that's true that would be ace. - Reduce my costs as well as clutter in the office. ;-)
12:21.17tgreerdacm_work: i may also be considering a purchase of them if they do work like that
12:21.21tgreerif you find out gimme a shout
12:21.32dacm_worktgreer: Will do.
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12:25.07stixWhere can I read about aliases setup in cdr_mysql.conf ?
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12:44.46Justmanhi guys
12:45.53Justmanhow i can limit incoming calls to my phone with asterisk 1.6/realtime postgres ? call-limit in the database table already set, but doesn't work :(
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12:56.14gonzhardthi
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13:04.29telnettechwhat is required to use XML scripting with Asterisk? this may be a dumb question but i havent used it this way before.
13:06.17[TK]D-Fendertelnettech: And what relationship for XML have with "Asterisk" ?
13:06.17Nuggettelnet is eeeeeeevil!
13:06.57[TK]D-Fenders/for/does
13:07.36jsmanwhere can I find more information about how custom cdr variables are passed to the cdr backends?
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13:08.53telnettechour old softswitch uses a XML file per customer for the purpose of caller ID, routing and other specific info that would pertain to that specific customer
13:09.25telnettechsince we have the files already, we were wanting to see if there was a way to use them in asterisk and not have to "reinvent the wheel"
13:10.14telnettechand move those customers over to an asterisk server with minimum disruption
13:10.39telnettechso that we can get rid of the old softswitch that is no longer supported by the vendor
13:10.49[TK]D-Fendertelnettech: XML is just an organization concept, not even a fixed syntax
13:11.07[TK]D-Fendertelnettech: that structure doesn't relate to dialplan at all.
13:12.12telnettechok so it is going to be the painstaking, time consuming task of building each customer into a context of their own...ugh
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13:14.29jsmanif i Set(CDR(foo)=abcd) can I retrieve the value for foo in cdr_radius for example?
13:15.31WIMPytelnettech: Why not write a converter?
13:17.29telnettechWIMPy: using what to convert from XML to Asterisk dialplan?
13:17.53kaldemartelnettech: easy as hell if you can write a little XSLT.
13:18.47WIMPytelnettech: Some language you're familiar with.
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13:20.21telnettechonly thing i know how to do, programming wise, is XML and bash scripting
13:21.00telnettechso it is possible to convert XML to say PERL or PHP and get it to work in Asterisk....using the AGI/AMI?
13:21.37telnettechor just converting to a format that you put into the dialplan file directly?
13:22.04WIMPyI'd suggest the later.
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13:24.28upbo m f g
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13:26.51upbtelnettech: what you could do is implement a 'realtime' dialplan extension or whatever it is in asterisk lingo, like
13:27.08upbthe ones that load parts of dialplan from rdbms
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13:29.06upbbut instead it would generate the dialplan from your xml
13:29.43telnettechok thanks guys....looks like im going to need to dive into this further
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13:48.20KattyGOOD MORNING ALL YOU BEAUTIFUL PEOPLE
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13:48.51b0ot[TK]D-Fender, can a phone register without a secret?
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13:50.56lirakisany one know if there is some documentation about the telnet interface on grandstream phones?
13:54.35SaiSomahey guys, I'm having an issue with allowing callers to exit a queue and be directed to voicemail.  I'm stumped:(
13:54.41SaiSomaHere's pertinent info: http://pastebin.com/Qeu4K4FB
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13:56.08pabelangerAnybody know of a Polycom phone that support Multicast RTP paging?
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14:07.24Naikrovekpabelanger: if one does, they probably all do, and I don't think any do.
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14:12.12jimi_How can I match all US toll free prefixes in 1 pattern?
14:13.19pabelangerwhy 1 pattern?
14:13.39anonymouz666let jimi take over
14:13.55tgreerthought all us toll free numbers were 1-800?
14:14.07pabelanger800, 866, 888, 877
14:14.23jimi_yeah, there are several
14:14.28tgreerheh
14:14.36jimi_pabelanger, 1 pattern, to keep the dialplan clean and manageable
14:14.40tgreercan tell i'm not american :)
14:14.41russand more reserved for future possible 800 use
14:14.56jimi_I tried like 18[866|877] etc but that didnt work
14:15.04jimi_whoops
14:15.12jimi_1[877|866] etc
14:15.17russjust make 4 patterns
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14:16.33jimi_russ, its like 15 patterns
14:16.38russwhat'll really blow your mind is when you discover that calls that start with 1 can call some really far away expensive places
14:17.12jimi_Also, we have 8 linked servers that we have to replicate this diaplan across
14:17.26russright now its only 800, 888, 877, 866, and 855
14:17.28pabelangerjimi_: Yes, more work but more secure.
14:17.33pabelangerjimi_: DUNDi
14:18.14pabelanger1 pattern match would be exten => _1X.
14:18.29pabelangeror _18.
14:19.19russwhy not _1NXXNXXXXXX?
14:19.53pabelangerthat would match all 10 digit numbers
14:20.21pabelangerthat is the downside
14:20.52russwell, _18. would not be a good pattern, it would match the Dominican Republic
14:21.55pabelangerruss: Agree, there is no easy method. I'm with your original statement, create a pattern for each
14:25.27jimi_Also 804 is an area code in VA
14:25.42russamong many others
14:27.51SaiSomahey guys, I'm having an issue with allowing callers to exit a queue and be directed to voicemail.  I'm stumped:(
14:27.53SaiSomaHere's pertinent info: http://pastebin.com/Qeu4K4FB
14:31.21jimi_In the asterisk CLI, how can I isolate the output to only 1 user's  or inbound #, etc?
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14:34.58Kattyblahs
14:35.24b0otwhat is a sip domain vs a sip regitar?
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14:36.42kjsanyone know of a decent UK voip ITSP ?
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14:53.13megalomanohi people , I have some doubts refers to dialplan an Goto - ${DIALSTATUS} ,,, for example , i have the next lines: XXXX,2,Goto(XXXX-${DIALSTATUS},1} ---- XXXX-BUSY,1,Goto(somextension)----- XXXX-CONGESTION,n,Goto(somextension) , here , dialplan only takes the first condition and the second is discarded ,, is right this sintax?
14:55.44tgreerand i'm off to do some work before i go home
14:55.48tgreerttyl all
14:56.15b0otI still have status UNKNOWN in sip show peers for a new sip client im trying to attach
14:56.40b0otI can make calls out from it
14:56.42b0otbut i can't call it
15:01.00dacm_workCould anyone here give me a rough idea of how much bandwidth a call (of the same quality as an analogue line) requires?
15:03.32deadpigeonis there any way to get smdi to work over IP links?
15:03.39p3nguinmegalomano: When you call an extension, a DIALSTATUS should be generated.  When that extension exits, you're matching a new extension which IS that DIALSTATUS status code.  What's the problem?
15:03.46deadpigeonEverything wants a rs232 interface...
15:04.04deadpigeonBut our coppercom switch does -everything- via IP when it comes to smdi
15:04.29fauxallianceSMDI wants a serial connection...
15:04.36mmatticetcp to serial isn't that difficult a conversion
15:04.40fauxalliance<EOT>
15:05.25deadpigeonI was hoping someone had encountered this before and wrote something up on it.. but after a few days of poking around I think I'm close to giving up.
15:05.34deadpigeonI just want a voice mail solution that isn't going to be costly..
15:09.44fauxalliancehttp://www.voip-info.org/wiki/view/Asterisk+bounty+SMDI
15:09.48fauxalliancecheap(er)
15:10.01megalomanop3nguin: that sencond condition is discarded ( first cond. XXX-BUSY........... second XXXX-UNAVAILABLE ) ,,, the line that is dialed could be in any state ,, BUSY or UNAVAILABLE ,, but asterisk only takes the firs (busy and the line is not available)
15:11.04p3nguinmegalomano: XXXX-BUSY is an extension.  XXXX-UNAVAILABLE is a different extension.
15:11.50p3nguinmegalomano: If the exiting line in the original Dial was to go to XXXX-${DIALSTATUS}, then it will only match whatever the DIALSTATUS is.
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15:12.18[TK]D-Fenderb0ot: Perhaps you should seeing why it isn't registering
15:12.22Tim_Toadymegalomano: the syntax should be like exten => XXX,n(BUSY),foo u should be using priority tags
15:12.59b0ot[TK]D-Fender, how do i see that?
15:12.59p3nguinmegalomano: If the DIALSTATUS was WTFISTHISSTATUS, then it would try to go to an extension matching XXXX-WTFISTHISSTATUS.
15:13.19[TK]D-Fenderb0ot: SIP DEBUG <-
15:13.42dacm_workDoes call diversion work at the IP-PBX level or upstream at the SIP Trunk provider?
15:14.37b0otsweet
15:15.38megalomanothanks =)
15:17.29Kattyyawns
15:18.15p3nguinmegalomano: You can use Verbose(${DIALSTATUS}) or NoOp(${DIALSTATUS}) in the dialplan to see the DIALSTATUS.
15:19.26Katty[TK]D-Fender: <3
15:20.06[TK]D-Fenderhugz on teh Katty
15:20.24Faustovyou're less than 3 ;(
15:20.46*** join/#asterisk moy (~moy@UNVLON55-1176057127.sdsl.bell.ca)
15:20.48[TK]D-FenderFaustov: My band is "Less Than 4" :p
15:21.21Kattythat's what she said.
15:21.29Faustovgood, means it can be coded on just 2 bits ;)
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15:22.28Kattyhi KavanS
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15:26.51megalomanoother hand is ringing all lines at the same time ,is possible to do this without queueu an agents?
15:26.59megalomanoand agents?
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15:27.34anonymouz66664 E1s in just one equipment
15:27.45anonymouz666that's a lot
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15:28.30[TK]D-Fendermegalomano: strategy=ringall
15:29.36*** part/#asterisk rossand (~aross@95.215.120.8)
15:30.41anonymouz666if you use ringall don't forget to alert the operators that some calls will be "mute" - that's because someone faster pickup first.
15:32.00megalomano[TK]D-Fender: but  the  strategy is used by agents and queues ,, right?
15:32.30[TK]D-Fendermegalomano: Strategy is used by QUEUES
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15:32.47[TK]D-Fendermegalomano: Agents don't run the show
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15:33.06fauxalliancenothing better than watching hundreds of agents reading a 'dead air "script"'...
15:33.19megalomano[TK]D-Fender: ok then , the queues not needs agents
15:34.00[TK]D-Fendermegalomano: queues need MEMBERS.  an "Agent" could BE a member
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15:35.03megalomanoD-Fener: ok , MEMBER could be a sip user ( registered into mi *)
15:35.27[TK]D-Fendermegalomano: Yes
15:35.37[TK]D-Fendermegalomano: NY channel type
15:35.41[TK]D-FenderANY
15:35.47*** join/#asterisk adyn (~adyn@unaffiliated/adyn)
15:36.02[TK]D-Fendermegalomano: Whatever you could put in a Dial() call
15:39.07asilvaHello guys.. i a little help... im using asterisk 1.6.2.11 i want to patch for 12-rc1 .. just download the patch and apply to asterisk source folder?
15:40.00asilva[TK]D-Fender, ?
15:40.37[TK]D-Fenderasilva: Do not target individuals for support like that.... and I don't do "patches"
15:40.55asilvanp
15:41.03Kattyi'll patch you in a minute.
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15:41.55datacompboyHi! Anybody knows where in zaptel source decode of input number over PRI line is done?
15:41.58asilva?
15:42.38megalomano[TK]D-Fender:for example , I have 4 sip clients , I have to add this clients as a member,after define their dialplan XXX,1,Queue(queue's name) ,,, right?
15:43.04[TK]D-Fendermegalomano: That's how Queue() works...
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15:46.32solherro :)
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15:50.30megalomano[TK]D-Fender:ok , excuse , but I have doubts refers to define members that are not Agents
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15:50.48[TK]D-Fendermegalomano: member=SIP/100
15:50.54[TK]D-Fendermegalomano: THERE
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15:52.56javierjhi there... I'm wondering how can I set polarity reversal detection for just one channel of my tdm24 card... any advise?
15:53.20[TK]D-Fenderjavierj: To do what?
15:54.02[TK]D-Fenderjavierj: * DOES detect it.  Its a question of what you expect it to do with it
15:54.15javierj[TK]D-Fender: I need to activate answeronpolarityswitch for just one of the fxo ports of a tdm card... is it possible?
15:54.33[TK]D-Fenderjavierj: If that is an option at all, then yes
15:55.34megalomano[TK]D-Fender: THANKS
15:55.59javierj[TK]D-Fender: How can I do that, when I set answeronpolarityswitch=yes it seems to apply for all the ports
15:58.44[TK]D-Fenderjavierj: You don't seem to understand how DAHDI confgis work.. this IS per-channel.
15:59.04ruyojavierj, before each channel declaration, set your answeronpolarityswitch option.
15:59.08[TK]D-Fenderjavierj: You on the other hand set all your parms and allocated them to ALL channels in something like "channel =>1-23
15:59.27[TK]D-Fenderjavierj: So set parms.  specify channels.  CHANGE some.  Specify OTHER channels
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16:00.41javierj[TK]D-Fender: thanks
16:00.54javierjruyo: thanks
16:02.00grayhamehello all.  i've had an asterisk pbx that is crashing from time to time.  this build doesn't have debugging enabled and I can't recompile it to enable it.  i've looked at asterisk/messages for hints but see nothing.  my question is, where else can I look to try and figure out why this system crashes?
16:02.42*** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb)
16:03.05ruyograyhame, try dmesg.
16:04.44grayhameruyo:  thanks so much
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16:07.45p3nguinmegalomano: You don't have to use queues to ring multiple phones.  Dial(SIP/tom&SIP/dick&SIP/harry) will work just fine.
16:10.59megalomanop3nguin: thanks :)
16:11.42megalomanop3nguin: at this sintax . I can add the dialstatus?
16:12.05p3nguinTry it and let me know.
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16:23.01heffertzafrir_laptop: thanks for the changes for 2.4.0 on your patches! unfortunately i now get "unable to register zaptel device!" in dmesg on my HFC-S using zaphfc. any thoughts?
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16:37.03megalomanop3nguin: only ringing one extension , and I have some doubts with priority and if make a travel through the lines availables
16:37.19p3nguinYou're doing it wrong.
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16:39.36erotavlas_turbohi
16:39.46erotavlas_turboI have a question
16:40.33*** part/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com)
16:40.53dadavoipand then the pause
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16:41.50erotavlas_turboHi
16:42.06dadavoipyou said that already Hi
16:42.16erotavlas_turboSorry, I'm new of IRC chat
16:42.27erotavlas_turboI have a question about Asterisk development
16:42.45dadavoipasterisk will do that make you cahnge your ways
16:42.56WIMPydadavoip: Wait and see if there will be "lo" as well. It could become binary then.
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16:43.21erotavlas_turboI would like to know an information about ast_channel
16:43.46WIMPyerotavlas_turbo: Just ask.
16:43.51WIMPy~ask
16:43.51infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
16:44.02dadavoiphttp://www.asterisk.org/doxygen/asterisk1.0/structast__channel.html
16:44.30erotavlas_turboIn version 1.6.2 the structure ast_channel has a member called ast_callerID, the new version 1.8 hasn't this structure
16:44.53leifmadsensounds like an #asterisk-dev question
16:45.08erotavlas_turboyes, I know the documentation links
16:45.19erotavlas_turbook, thank you
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16:55.44b0ot[TK]D-Fender, p3nguin any idea what is wrong with 10.1.40.100 ? http://pastebin.com/daRpeGnS
16:57.40dadavoipdo you have trouble registring the channel?
17:03.03*** join/#asterisk gr0mit (~tim@eastrop.plus.com)
17:04.20b0otdadavoip, I just can't get the client to register
17:05.53dadavoipb0ot lam wan?
17:05.57dadavoiplan
17:06.38b0otlan
17:07.11b0otim able to call out from the phone, but it's not registering
17:07.11Kattyomnomnom tuna salad
17:07.16b0otor able to recieve calls
17:07.56dadavoippaste cli sip show peers and cli of outgoing call
17:08.17[TK]D-Fenderb0ot: "sip show peer [thepeer]
17:08.38p3nguinb0ot: Not registering and not able to receive calls usually goes hand in hand.
17:08.41dadavoipapple
17:09.01dadavoipp3nguin:thought so
17:09.21*** join/#asterisk minaguib (~mina@modemcable109.56-20-96.mc.videotron.ca)
17:09.54p3nguinSince, you know, registering tells Asterisk HOW TO REACH THE DEVICE when it has a call for it.
17:10.15Katty^- win.
17:10.16minaguibI'm trying to set up some sort of hotline where *someone* must be reached. Currently worried that if the outgoing call is picked up by a cell phone's company (voicemail/out of service/etc) the call will end there.  Any way I can dial (in parallel or in sequence) a list ansuring a human answers ?
17:10.46Kattyyou could do find me follow me thing
17:11.06Kattyjust make sure the rings are less than what's required to voicemail to pick up
17:11.14b0ot[TK]D-Fender, p3nguin dadavoip sip show peers http://pastebin.com/Gmw9vjYH
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17:11.37minaguibKatty: That's variable. For example if a cell phone is OFF, the company will pick up after 1 ring..
17:11.39Kattyfollow me is kind of annoying tho
17:12.13dadavoipp3nguin: asterisk sucks there is no plug and play(future releases hopefully before version 2.2176 would include an application called dummy that will do all of this for u) you just sit back and relax and bobs your uncel, working voip
17:12.36Kobazdadavoip: plug and play voip?
17:12.37Kattyminaguib: well maybe someone else will have an idea for you--that's all i got
17:12.39[TK]D-Fenderb0ot: Status       : UNREACHABLE: You have a LOT of retries.  What networking is between your server and that device?  What IS that device?
17:12.43Kobazdadavoip: how's that supposed to work?
17:13.03minaguibIs there something that perhaps requires the dialed number to say "Answer/Accept" ?
17:13.12[TK]D-Fender[13:10]<minaguib>I'm trying to set up some sort of hotline where *someone* must be reached. Currently worried that if the outgoing call is picked up by a cell phone's company (voicemail/out of service/etc) the call will end there. Any way I can dial (in parallel or in sequence) a list ansuring a human answers ? <- Dial() + (M()
17:14.14p3nguindadavoip: Well, that does exist, but it'll cost you at least $50 per hour.
17:14.22Katty[TK]D-Fender: i did 3 sets of pushups at the gym today
17:14.26Katty[TK]D-Fender: girly ones, but still push ups!
17:14.30Kobaznice
17:14.35Kobazi've been doing this abs circuit
17:14.39b0ot[TK]D-Fender, it is just a switch
17:14.43b0oton the same subnet
17:14.51b0otbetween a laptop running asterisk and my sip client
17:15.04Kobazkatty: 30 seconds of crunches, 30 seconds of bicycle, 30 seconds of scissors, and 30 seconds of the plank... and do two sets
17:15.07KattyKobaz: oh yeah? what does the circuit include?
17:15.24Kattydang
17:15.27Kobaz30 seconds rest in between sets
17:15.29Kattyi care barely hold plank alone for 30 seconds
17:15.31b0otit's essentially a softphone
17:15.43[TK]D-FenderESSENTIALLY?
17:15.50[TK]D-Fenderb0ot: Please get specific FAST
17:15.55dadavoiplol it either is or its not
17:16.09[TK]D-Fenderb0ot: Vague shit and disappearing packets is not makign debugging this very appealing
17:16.15Kobazessential... like essential oils that smell oh so lovely
17:16.36[TK]D-Fenderlights Kobaz's essential oils on FIRE
17:16.37dadavoipekiga running on xt pc with car batery to keep up with bios mods
17:17.29ectospasmminaguib: you can play a message to what answers, and they have to respond with a DTMF digit.  If they don't respond appropriately, do something useful with the call.
17:17.34drmessanoI thought essentially was a send confirmation
17:17.40*** part/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
17:17.48b0ot[TK]D-Fender, I have a laptop 10.1.40.1 (ubuntu 10.04) running asterisk it is connected to a switch which is connected to a another linux box running a softphone
17:17.51Katty[TK]D-Fender: get a room for the kinky stuff
17:17.55drmessano"Did you get his email?"  "he sent, surely?"
17:17.58*** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net)
17:18.02ectospasmminaguib: a telco message won't hit the right DTMF like a live person would
17:18.14[TK]D-Fenderb0ot: I'm seing retransmits.  You eather have a firwewall issue, or a networking issue.
17:18.24Kattyor both
17:18.27[TK]D-Fenderb0ot: that aside it IS registered.  Jsut disable QUALIFY for your peer
17:18.28minaguib[TK]D-Fender: You mean dial with M(some-macro) for the called person ?
17:18.43[TK]D-Fenderminaguib: Yes.  Prompt for CONFIRMATION.
17:18.51drmessanoYou have a laptop on 10.1.40.1 running 10.04.  Fscking showoff.
17:18.56[TK]D-Fender[13:17]<ectospasm>minaguib: a telco message won't hit the right DTMF like a live person would <- as such
17:19.09dadavoiplolololololooololol
17:19.21minaguibCool.  Does asterisk offer something I can use or I' have to write the macro from scratch ?
17:19.34ectospasmminaguib: use the READ application
17:19.37Kattyecto...spasm.
17:19.40Kattysounds soooo wrong
17:20.04ectospasmKatty: it's in the Urban Dictionary, but I assure you I arrived at the nick independently
17:20.05[TK]D-Fenderminaguib: All dialplan = YOUR job
17:20.10dadavoipsound sore
17:20.11[TK]D-Fenderminaguib: You should know this already
17:21.13Kattyectospasm: s'all good
17:23.14Kobazi gots a good question
17:23.42[TK]D-FenderI'm sure you does
17:24.02Kobazif you do ChanIsAvail on a local channel... is it guaranteed that if you then dial that local channel... that's the channel that gets used?
17:24.03Kattyi gots a good answer
17:24.05Kattythink they match?!
17:24.18Kattythey don't :<
17:24.31QwellKatty: I have a question that would match!
17:25.26[TK]D-FenderKobaz: pardon?  A local channel only shows the availability of the DIALPLAN behind it.
17:25.36[TK]D-FenderKobaz: it isn't a port to any other specific tech
17:25.45Kobazbut like
17:25.48b0ot[TK]D-Fender, how do you disable qualify for a peer
17:25.57Kobazyou get the   AVAILCHAN     - the name of the available channel, if one exists
17:26.00[TK]D-Fenderb0ot: qualify=no <- duh
17:26.04b0otand the're aren't firewalls in linux are there
17:26.19b0ot[TK]D-Fender, the only option was sip qualify peer kdrto
17:26.22Kobaz[TK]D-Fender: it'll set AVAILCHAN to be like Local/foo@bar-223487 etc
17:26.23dadavoiplol
17:26.37Kobaz[TK]D-Fender: is that a stable value of the channel-to-be-used
17:26.37[TK]D-FenderKobaz: It will return a valid DEVICE.  You are testing a DEVICE to be available for issuing a CHANNEL.
17:26.39b0ot[TK]D-Fender, or do i put that in sip.conf?
17:26.49Kobazyes i know it's a device... but the app sets AVAILCHAN
17:26.59[TK]D-Fenderb0ot: You PEER is in sip.conf
17:27.01Kobazwhich looks to me... like it could be a channel that might get used
17:27.12[TK]D-FenderKobaz: Hell no <-
17:27.18Kobazblarg
17:27.22[TK]D-FenderKobaz: it returns which DEVICE was availabl
17:27.36Kobaz[TK]D-Fender: it also sets a channel... lemme show you
17:27.37[TK]D-FenderKobaz: You are NOT going to get uniqueID usable shit out of that
17:27.41*** join/#asterisk n3hxs (~HAMming@63.68.135.4)
17:27.48Kobazk
17:27.53Kobazthat's what i was asking... heh
17:28.44Kobazi kind of want the channel that's going to be used before i use it
17:28.51*** join/#asterisk gr0mit (tim@eastrop.plus.com)
17:29.00Kobazi can get the channel with a pickup macro... but then i might incude race conditions
17:29.04Kobazinduce
17:30.26p3nguinIs it possible to force a uniqueID before the call begins?
17:30.49dadavoipyes
17:30.53Kobazreally?
17:31.09p3nguinIf so, that's how you can do whatever it is that you're trying to do.
17:31.42Kobazdadavoip: how's that work?
17:32.08Kobazthat would totally solve all kinds of quirks i have with existing apps
17:32.36dadavoipKobaz:sorry cught the tail of conversation what do you want to do?
17:32.46Kobazit would be nice if you know you want to dial sip/foo  or local/a@b... and you could reserve the channel name... and then say okay... go do it... kind of like a prepared dial
17:33.49KobazPrepareDial(SIP/foo,options..etc)  Set(foo=${PREPARED_CHANNEL})  DoPreparedDial()
17:33.52Kobazsomething like that
17:34.07dadavoipchanisavail
17:34.20*** join/#asterisk ccesario_ (~ccesario@189-29-32-71-ac.cpe.vivax.com.br)
17:34.23Kobazyeah but fender just said the channel isn't stable
17:34.56[TK]D-FenderKobaz: It doesn't return a channel.  It tests the device to see if it is conceivable to open ca channel against it
17:35.06Kobazyes... and it also sets AVAILCHAN
17:35.09Kobazlemme do some tests
17:35.09dadavoip0 or 1
17:35.18dadavoipyes or no
17:35.19Kobazwhich *looks* like a channel
17:35.20[TK]D-FenderKobaz: which is a DEVICE NAME, not a CHANNEL
17:35.44[TK]D-Fenderreaches for his rusty-nail upgraded ClueBat (tm)
17:35.51anonymouz666SPA3102 is useless for FXO. The echocan just doesn't work.
17:36.04[TK]D-Fenderanonymouz666: It does.  Not always stellar.
17:36.13[TK]D-Fenderanonymouz666: Also depends on which firmware VS your conditions.
17:36.21anonymouz666latest firmware
17:36.38[TK]D-Fenderanonymouz666: DOWNGRADE or play with related settings.
17:36.47Kobazchanisavil returns a different channel for each call
17:37.11*** join/#asterisk QubeZ (~nkasu@64.128.254.34)
17:37.14QubeZhello all
17:37.14*** join/#asterisk gr0mit (~tim@212.56.94.150)
17:37.17anonymouz666[TK]D-Fender: very tired to play with gains... works sometimes and then the echo backs.
17:37.21Kobaz<PROTECTED>
17:37.24Kobaz<PROTECTED>
17:37.38Kobaz- SIP/201-00000134 is ringing
17:37.40dadavoippaste that sh@t
17:37.42QubeZi have a queue setup with 4 members - 2 of which have higher penalty. When I ring into the queue, it bounces only between the first two members that have no penalty. Queue is rrmemory.
17:37.46Kobazsoooo
17:38.01Kobazchanisavail just makes random channel names but they never get used
17:38.24Kobazthe noop is the AVAILCHAN from ChanIsAvail
17:38.26anonymouz666some conditions just works and some doesn't. as you can't guess if will work on the customer line, it's better to consider another option
17:38.27dadavoiplol some please send the link for the latest manual
17:38.58Kobaz[TK]D-Fender: dadavoip: so... i think what I'm asking for is what i mentioned earlier... like a prepared dial
17:39.43[TK]D-FenderKobaz: "Do or do not, there is no try" </yoda>
17:39.43Kobaz[TK]D-Fender: ChanIsAvail does indeed set a channel... but it's not a channel that means anything
17:39.53Kobaz13:39 -dadavoip(~d@41-134-22-10.dsl.mweb.co.za)- DCC Chat (41.134.22.10)
17:39.55Kobazwhat's what for?
17:40.39dadavoipwanna chat
17:40.50Kobazasl?
17:41.09dadavoip22 f zulu
17:41.27Kobazneat
17:41.35Kobazso how bout them mets
17:41.44*** join/#asterisk d-tech (~d-dtech@72.245.233.107)
17:41.54dadavoipi have just the code for you Kobaz tat will work
17:42.04Kattyi wanna chat.
17:42.08Kobazokay
17:42.12Kobazthat would be cool
17:42.16Kobazi'll bake you some cookies
17:42.25Katty<3
17:42.34dadavoipexten =>_X.,1,Goto(bookstore and get the manual)
17:42.48Kattycookies don't need a manual.
17:42.49Kobazdadavoip: there is no manual for this
17:43.04Kobazdont be a smartass
17:43.18dadavoiplol sry i thought i was in the asterisk channel
17:43.41upbwtf :D
17:43.55SaiSomahey guys, I'm having an issue with allowing callers to exit a queue and be directed to voicemail.  I'm stumped:(
17:43.57SaiSomaHere's pertinent info: http://pastebin.com/Qeu4K4FB
17:43.57Kobaz[TK]D-Fender: time to add more code to asterisk :P
17:44.48Kattywhat? asterisk? here?
17:44.53Kattysurely you're joking
17:45.38*** join/#asterisk darksk1ez (~mhb@darkskiez.ipv6.darkskiez.co.uk)
17:45.45minaguibSo, I was hunting the existing voice promtps for "accept or reject" for my macro
17:45.53minaguiband I discovered that FindMe does exactly what I want !
17:46.09[TK]D-FenderSaiSoma: Problem I see is nowhere in there do you make ANY reference to Voicemail()
17:46.18minaguibFindMe(findme_id), and in findme.conf I define that id and who it calls
17:46.46SaiSoma[TK]D-Fender: that's further along in the dialplan. hang on and i'll post more (didn't think it was needed, but I'm stoopid sometimes;) )
17:48.05minaguibIs there a reason why I should code my own macro instead of using FindMe ?
17:48.24SaiSoma[TK]D-Fender: here's the other pieces that should be relevant: http://pastebin.com/FCPz3xgq
17:48.58[TK]D-FenderSaiSoma: Better control over what gets called and when.  And how.
17:49.31zxvffgood morning [TK]D-Fender, my pbx works today!
17:49.45zxvffyou know, just in case you were hoping for an update
17:49.49[TK]D-FenderSaiSoma: * just quits the QUEUE.  After that you have 1 playback and your acll dies.
17:50.06[TK]D-FenderSaiSoma: You aer not hitting VM after your queue exits in-line
17:51.01[TK]D-FenderSaiSoma: And I fail to see the relevance on that lower context in what you've shown.
17:51.10SaiSoma[TK]D-Fender: holy cow I missed four lines in the [helpdesk-queue] context.  gah.  been a bad day.  reposting:(.  sorry.
17:51.58anonymouz666[TK]D-Fender: there's also a bizarre situation that you are speaking and studdenly another call appears at the same time as the current one
17:52.06SaiSoma[TK]D-Fender: OK.  Here, it's right this time. Sorry for wasting your time:(.    http://pastebin.com/zRynBa4s
17:52.10*** join/#asterisk rrb3942 (~rbullock@208.34.105.161)
17:52.53*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
17:54.30[TK]D-FenderSaiSoma: Show me the COMPLETE failed call
17:54.54[TK]D-Fenderminaguib: [13:48]<[TK]D-Fender>SaiSoma: Better control over what gets called and when. And how.
17:54.58[TK]D-Fenderminaguib: was for yuo
17:55.01p3nguinAs long as your context in queue.conf is that of the helpdesk-queue, pressing 1 while in the queue might take the call to vm.
17:55.32SaiSoma[TK]D-Fender: *nod* and I'll add further explanation  one min
17:55.51*** part/#asterisk C4colo (~DJpyro@184-96-209-31.hlrn.qwest.net)
17:57.37*** join/#asterisk Ryushin (proxy@windwalker.openinnovations.com)
18:00.55SaiSoma[TK]D-Fender: ok.  here it is: http://pastebin.com/TUtRh3S7
18:01.13SaiSomaif you need any other items from the dialplan, i can post, but I don't think they are relevant.
18:01.44SaiSomaThe queue acts as I expect (re-ringing phones every 25s), except that the caller cannot break out to leave a voicemail
18:02.12*** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec)
18:02.12[TK]D-FenderSaiSoma: And how are you trying to "breah out"?  H KILLS THE CALL
18:02.26[TK]D-FenderSaiSoma: context= <-- what you should be using
18:02.52p3nguinIt's like words I type don't print on the screen or something.
18:03.28*** join/#asterisk marcompile (~marco@189.12.200.32)
18:04.07marcompilehello, I have an AGI script that takes some time to load. Is there any way I could have a pool of started scripts waiting for asterisk input?
18:04.10SaiSoma[TK]D-Fender: context= is in queues.conf.  context=helpdesk-vm  I thought (though may very well have misunderstood) that pressing * while in the queue would allow the caller to exit the queue and return to the context prior to timeout.
18:04.26marcompileso I wouldn't have to wait for the scripts to load every time I call them?
18:04.35Kobazmarcompile: you can... that's more of a language specific thing. look into fastagi
18:04.40[TK]D-FenderSaiSoma: NO.  "H" and "context=" aer OPPOSING ideas.
18:05.01[TK]D-FenderSaiSoma: Do NOT do "H".  Juse put SINGLE DIGT extens you want it to be able to quit to.
18:05.08Kobazmarcompile: fastagi will do a tcp socket connection to your already running agi daemon... how you implement that daemon is up to you
18:05.25p3nguin; A context may be specified, in which if the user types a SINGLE
18:05.25p3nguin; digit extension while they are in the queue, they will be taken out
18:05.25p3nguin; of the queue and sent to that extension in this context.
18:05.37p3nguin(1255.01) <p3nguin> As long as your context in queue.conf is that of the helpdesk-queue, pressing 1 while in the queue  might take the call to vm.
18:05.44Kobazmarcompile: keep in mind that the agi daemon you write will need to handle multiple connections, and multiple things going on at the same time, if you plan on throwing more than one call at a time to it
18:06.00SaiSoma[TK]D-Fender: I think I am beginning to understand. So in queues.conf, put context=helpdesk-vm (as it is) and have an extension in that context that the user dials (* or whatever).  Am I understanding?
18:06.14[TK]D-FenderSaiSoma: yes
18:06.17p3nguinJesus.  Pay attension.
18:06.46p3nguinIf you have context=helpdesk-vm in your queue conf, you're going to be screwed because you can't dial 's' on your phone.
18:07.19marcompileKobaz, oic
18:07.23marcompileKobaz, thanks for the info..
18:09.01SaiSoma[TK]D-Fender: as always, you guys rock.  Sorry for my misunderstanding.  Thank you!
18:10.04SaiSomap3nguin: my apologies as well.  I obviously misread that as I was expecting something else (no idea why).  Again, thanks:)
18:11.36*** join/#asterisk Benwa (~Benwa@unaffiliated/benwa)
18:13.02*** join/#asterisk Mhaddog (~Mhaddog@z65-50-116-17.ips.direcpath.com)
18:15.42*** join/#asterisk odenkos_ntbk (~odenkos@ip-212-081-019-170.static.nextra.sk)
18:17.36*** join/#asterisk b0ot (~Jinxed---@147.177.40.62)
18:18.08b0ot[TK]D-Fender, Can your recommend a nice softphone that is easy/reliable to set up with asterisk
18:18.19b0otmy SJphone isn't able to talk
18:18.30b0otbut it can recieve audio just fine
18:18.30bougymanwe use SJphone with asterisk fine.
18:18.39bougymanthough we're upgrading to eyebeam
18:18.48bougymansjphone hasn't had any dev on it in a few years.
18:18.51b0otwell they were working this morning
18:18.54b0otand now they aren't
18:18.59b0otseems unrelible
18:19.12bougymanwe've had it at a call center for 9 months with no issues.
18:19.27bougymanagainst asterisk 1.4.23 + orderly stats
18:20.37*** join/#asterisk neurosys (~neurosys@173.200.200.185)
18:20.40[TK]D-Fenderb0ot: and I have no reliable backup to say that it is at fault
18:20.59[TK]D-Fenderb0ot: And you didn't specify PLATFORM either
18:21.17bougymandid any routing change, b0ot ?
18:21.29bougymanone-way audio is almost always a networking/routing issue.
18:21.38bougymani assume nothing is natted?
18:21.45b0otno it's all a flat network
18:21.52b0otall on 10.1.40.x network
18:22.31b0otone laptop is 1 switch away
18:22.34bougymanyou try any other phone to see if it replicates the behavior?
18:22.37b0otone laptop is 2 switches away
18:22.52bougymantry xlite or 3cx, see if they behave the same.
18:25.00b0otnevermind im an idiot
18:25.03*** join/#asterisk Letoric (~Letoric@tpi-dfw-uc-f1-69-94-238-22.totalprocess.net)
18:25.06b0oti had both laptops as the same user
18:25.17b0otso it kept going back and forth
18:25.43b0ot:)
18:26.09LetoricI am trying to accomplish having 1 line on multiple phones, with all of the phones holding that line ringing at the same time. How best can I achieve this? We do not want it to ring the main line of the phones, as we want users to be able to set a separate ring for it, and also be able to forward their main line at their convenience.
18:26.17LetoricAnybody able to help with this?
18:26.26Letoricwe're using SIP
18:27.29*** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk)
18:29.27p3nguinYou want the best way?  Don't use one "line" on multiple phones.  Give every phone its own "line," and Dial() all of them when you want all the phones to ring.
18:31.01[TK]D-FenderHow do you "hold" a  ringing line?  What is that supposd to mean?
18:31.15Letoricthanks p3nguin
18:31.27Letoricis it not possible to achieve what I was trying for?
18:31.37p3nguinNot with Asterisk alone.
18:31.39Letoricseems like an operator phone would have everybody's line on it
18:31.46[TK]D-FenderLetoric: Your description is broken.  Please rephrase
18:31.49Letoricahh, figured that would be the catch ;(
18:31.55SaiSomaLetoric: what you are asking for is like the old "key system" I think.
18:32.19SaiSomaline 1 is telephone number 555-1212 or extension 9999
18:32.27p3nguinBut it's more sensible to Dial(SIP/jack&SIP/jill&SIP/bob&SIP/tom) anyway, in my opinion.
18:32.50p3nguinYou can easily have a single extension dial all those phones.
18:32.57Letoricyeah, we've been doing that
18:33.15LetoricI was just trying to improve on it by having 1 line on multiple phones, rather than have 1 extension dial multiple lines
18:33.27p3nguinThat's not an improvement.
18:33.32Letoricheh, ok ;)
18:33.37p3nguinEvery phone should have its own "name."
18:33.46Letoricwell, they do - but this was a 2nd line
18:33.54p3nguinunneeded
18:34.16Letoricand the only reason they have the 2nd line is because it's annoying for the users to be unable to separate ring tones/forward their lines when everything rings their main line
18:34.40[TK]D-FenderLetoric: What your phone SOUNDS like when it rings is ANOTHER matter
18:34.49p3nguinDoes the phone accept a distinctive ring command?
18:34.50[TK]D-FenderLetoric: You ahven't bothered telling us what you are USING.
18:35.13LetoricFender: We're using Asterisk 1.6.2.11 with Polycom SoundPoint IP670 phones
18:35.28[TK]D-FenderLetoric: then use a SIP HEADER to set the ringer to use <-
18:35.37p3nguinSo modify the "ringall" extension to change the ringer header.
18:35.50p3nguinerr, uh, SIP header
18:35.53Letoricok, but how would my users set this themselves?
18:36.06Letoricthat's what I was trying to achieve, user control of basic functions like ring tones and forwarding their line
18:36.27[TK]D-FenderLetoric: what is there to "set"
18:36.39[TK]D-FenderLetoric: WHY would your behaviour be changing?
18:36.40Kattyfender
18:36.53Kattyi'm having difficulty pronouncing 'me diraient en chour d'amiables chanson'
18:37.07Kattymi deray don cur de map eh shawnson?
18:37.34LetoricFender - when a call comes in currently, and somebody presses 1 to get to the helpdesk, that then rings multiple phones main line. Because of that, those users can't forward their main line, and they can't set a separate ring tone so they know when a call is coming to them, or to the helpdesk.
18:38.06p3nguinIf you want to know the target of the call, change the callerID information.
18:38.09Letoricthe device is capable of setting these things, however, it is only capable of setting it on lines it is aware of
18:38.28Kattywhere is junky, junky would know
18:39.20Kattywho speaks french
18:39.53n3hxshttp://translate.google.com/translate_t?hl=en&client=firefox-a&hs=Quk&rlz=1R1GGGL_en___US356&q=%27me+diraient+en+choeur+d%27amiables+chanson&um=1&ie=UTF-8&sa=N&tab=wT#fr|en|%27me%20diraient%20en%20choeur%20d%27amiables%20chanson
18:39.56n3hxsKatty
18:40.07n3hxsPush the Listen button.
18:40.26Kattypushes
18:40.46Kattynothing happens :<
18:40.52n3hxsMine doesn't speak :(
18:41.14Kattyi'm having extreme difficulty with d'amjables
18:41.14p3nguinOH NO, you've just activated the launch sequence!
18:41.22n3hxsMaybe it is tongue tied
18:41.59Kattyfile: do you speak french
18:46.22*** join/#asterisk chasing`Sol (~rc4@smtp.master-zone.net)
18:53.14citywokis there any documentation on realtime extensions in 1.6?  voip-info seems to be about it and they are talking 1.4
18:53.18mmlj4ich bin parle bitte francais, da
18:55.04[TK]D-FenderKatty: PM
18:57.25[TK]D-Fendercitywok: perhaps little has changed
18:58.47Katty[TK]D-Fender: i found junky
18:59.14*** join/#asterisk Trox (~Adium@85-126-174-18.work.xdsl-line.inode.at)
18:59.50[TK]D-FenderKatty: Pass on a "salut mon ostie!" for me wouldja? ;)
19:00.31Kattyk
19:00.48*** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec)
19:00.59*** join/#asterisk JunK-Y (~junky@64.15.77.94)
19:01.03JunK-Y[TK]D-Fender: :)
19:01.19[TK]D-FenderJunK-Y: Quoi de neuf?
19:01.30Kattywhat's neuf mean
19:01.58[TK]D-FenderKatty: "What's new"
19:02.02*** join/#asterisk lvlolvlo (~lvlolvlo@unaffiliated/lvlolvlo)
19:02.10Kattyk
19:02.16Kattyand ostie?
19:02.35[TK]D-FenderKatty: Best translation "'sup fucker?"
19:02.45Kattyoh i see.
19:02.47[TK]D-Fender(in a friendly way)
19:03.03Kattygathered as such
19:03.52citywok[TK]D-Fender: wrong, i found a pretty massive difference by googling for a while. they did change from | to , and it's not backwards compatabile.
19:04.21p3nguin1.4 should have been using , instead of | anyway.
19:04.27p3nguin(I think)
19:04.33citywokrealtime used | in 1.4
19:04.36[TK]D-FenderKatty: "mon" is possessive in a way that makes the negative term personal/friendly kinda way.  As oppoed to the fucker you distance yourself from :)
19:04.47p3nguinI guess that part never got updated in 1.4, then.
19:05.16[TK]D-Fendercitywok: 1.6 ditched "|" period
19:05.20citywokp3nguin: yea.  this is what i'm going off of: http://lists.digium.com/pipermail/asterisk-dev/2008-May/033096.html
19:05.22[TK]D-Fendercitywok: not jsut realtime <-
19:05.29*** join/#asterisk TimeRider (~steve@82.132.248.218)
19:06.11[TK]D-FenderKatty: So you good with your previous translation?
19:06.31Katty[TK]D-Fender: yeah JunK-Y told me how to pronounce it
19:06.59[TK]D-FenderKatty: Voulez-vous coucher avec moi ce soir?
19:07.01[TK]D-Fender:D
19:07.44Kattyummm
19:07.44Kattyno
19:07.45*** join/#asterisk ahowlader (~Adnan@119.30.38.77)
19:07.58Kattynow JunK-Y on the other hand, yes ;)
19:08.29anonymouz666[TK]D-Fender speaks french by default
19:08.54Kattyhmmhesays: no
19:09.02Kattyhmmhesays: do you just invite all your friends?
19:09.05[TK]D-FenderKatty: He's taken already, and she'd annihilate you :p
19:09.19Katty[TK]D-Fender: who said anything about competition?
19:09.30Katty[TK]D-Fender: i'd nap with her too
19:09.35[TK]D-Fenderanonymouz666: "by default"?  Not sure hwhat way you meant that..
19:09.44fauxalliancehe's frenchie
19:09.55anonymouz666[TK]D-Fender native language
19:09.56Qwellfauxalliance: don't let the french hear you say that ;)
19:09.59[TK]D-FenderMe?  No.
19:10.11JunK-Y[TK]D-Fender: ca va, ca va, toi?
19:10.20JunK-Y[TK]D-Fender: katty is for me, not you!
19:10.29fauxallianceQwell, I'm on an island... and speaking of the French, Indians and annihilation.
19:10.30Qwellsteals Katty from JunK-Y
19:10.38[TK]D-FenderMy mother tongue is Enghlish, but one should be default be decent in french if not fluent jsut for having lived here your whole life.
19:11.08[TK]D-FenderJunK-Y: Careful.... I still have that picture ;)
19:11.10fauxallianceone more frog gone...
19:11.10anonymouz666Katty: do you know spanish ?
19:11.30anonymouz666Katty: speak I mean.. it is a nice language very flexible
19:11.35JunK-Y[TK]D-Fender: which one?
19:11.41fauxallianceone of the romance ones....
19:11.44[TK]D-FenderI am a cunning linguist ;)
19:11.47*** join/#asterisk iratik (~itariki@74.223.41.171.nw.nuvox.net)
19:11.49Kattyanonymouz666: gato con queso
19:11.54fauxalliancenotes his cunning lingual skills....
19:11.56fauxallianceDAMNNIT
19:11.59anonymouz666hah
19:12.10[TK]D-FenderJunK-Y: You know which one.. with you & Leif post-pool after astricon :0
19:12.26*** join/#asterisk Tech_Travis (~Travis@wsip-64-58-136-162.oc.oc.cox.net)
19:12.27*** join/#asterisk s0me0ne (~kvirc@cpc1-walt6-0-0-cust855.13-2.cable.virginmedia.com)
19:12.30[TK]D-Fenderactually, wasn't Astricon.. one of the "get to Know Asterisk" lower level ones...
19:12.42Kattyooh la la
19:12.44Kattyi wanna see that photo
19:12.51JunK-Yhaaa, my ass? :)
19:12.55iratikDoes anyone out there know of any user friendly ways of editing the queue_log ? (A new software package we are using for queue reporting uses queue_log for tracking pause/unpause time .. we need the ability to have managers edit the pause/unpause times)
19:13.01[TK]D-FenderJunK-Y: INDEED :p
19:13.11JunK-YKatty: then you're owned by [TK]D-Fender now!
19:13.13JunK-Yhghahaahahhahah
19:13.20[TK]D-Fenderiratik: vi queue_log
19:13.26iratik[sic]
19:13.30Kattyi am not owned by anyone, kthx.
19:13.40[TK]D-FenderKatty: How much to rent?
19:13.42Kattyhowever i might be okay with occasionally being borrowed
19:13.50Kattyyou can't afford me.
19:14.02fauxalliancekudos
19:14.15[TK]D-FenderKatty: So now we've established your character and are just haggling ;)
19:14.22Kattybesides Qwell would get jealous.
19:15.04Kattyi'm still trying to weasel him into a date in ER
19:15.10fauxalliancethe difference between theory and reality....
19:15.17fauxalliancewinks at n3hxs
19:15.27QwellKatty: if you were here :p
19:15.38fauxalliancenot n3hxs ... sorry
19:15.47KattyQwell: minor detail.
19:15.50Qwellquite
19:17.13JunK-Y[TK]D-Fender: Katty : are ya comin to astricon this year?
19:17.53Kattydoubtful
19:18.24beardyWhere is it?
19:18.33JunK-Yastricon.net
19:18.48[TK]D-FenderJunK-Y: If its more than 1 province away or south of the border... nope
19:19.41[TK]D-FenderAll BOF Sessions   <-- sounds dirty
19:20.11[TK]D-FenderJunK-Y: Couldn't validate the trip
19:20.37[TK]D-FenderJunK-Y: that kind of $ gets me to the Bahamas for some R&R :)
19:26.16tzangernifty, zarlink makes some low-density echo can ICs
19:28.30pabelanger[TK]D-Fender: Drive, that is what I am doing
19:28.33*** join/#asterisk TheDavidFactor (~chatzilla@c-68-34-116-180.hsd1.md.comcast.net)
19:28.58[TK]D-Fenderpabelanger: From whre?
19:29.56pabelanger[TK]D-Fender: Ottawa
19:30.24[TK]D-Fenderpabelanger: Well.. still van't validate the $ and actually haven't dealt with getting my passport yet...
19:31.28*** join/#asterisk dangerarea (~kevin@kingfisherwalk.demon.co.uk)
19:32.15dangerareaevening all
19:32.51dangerareai'm wondering if something's possible and my trail of posts from google searches hasn't hit anything
19:34.29pabelanger~ask
19:34.30infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
19:34.55dangerareaif i have a pair of asterisk servers, can I connect a piece of ISDN kit to one and dial a data call through that to the other server and have the second dial the call
19:35.02dangerarea(sorry was trying to phrase that)
19:36.26dangerareai.e. if i have a server sat with connectivity and a mobile server to take out with me, can I plug some kit in that requires an ISDN B channel to make a data call through the mobile server over IP to the second and then have that data call fall out of the second server either to a genuine ISDN 2 line / another piece of equipment?
19:37.07*** join/#asterisk ccesario_ (~ccesario@187.75.139.188)
19:37.56[TK]D-Fenderdandre: Where are these 2 servers located relative to each other?
19:38.04hmmhesaysKatty, Negative
19:38.15hmmhesaysI had select them
19:38.24hmmhesays*hand
19:39.33dangerareathe servers would be over the internet but would be vpned through for as good a connection as I could get
19:40.36[TK]D-Fenderdangerarea: Well either of your servers can process whatever calls in whatever way you set them up to.
19:43.26dangerarea[TK]D-Fender: see I thought that would be the case but I've not seen anything relating to making 64/128k data calls through another server
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19:44.01dangerareaalso, what kind of interface cards would I require to conect 2 pieces of equipment back to back in that scenario (i.e. there are no genuine lines involved)
19:44.14*** join/#asterisk Navion (~Bill@rivendell/users/navion)
19:44.33NavionAny radio broadcast people here?
19:44.34[TK]D-Fenderdangerarea: You just mentioned they would be networked together over VPN.  This seems to imply VoIP between them  You are not running ISDN between them
19:45.13dangerareasee, this is what i'm trying to do
19:45.22dangerareaessentially extend ISDN over IP
19:46.15[TK]D-Fenderdangerarea: your ISDN has NO relation to VoIP between your servers
19:47.11[TK]D-Fenderdangerarea: You aren't "extending" anything.  * gets a call.  that is one leg.  * then calls out.  that is another leg. that is all.
19:47.40[TK]D-Fenderdangerarea: If you use voip between your servers taht does no imply the same bandwidth as normal ISDN might use.
19:47.57[TK]D-Fenderdangerarea: These calls are merely bridged by the connecting server.
19:48.22[TK]D-Fenderdangerarea: Your VoIP call will use whatever protocol & codec you set it up to use and the requirements will vary accordingly
19:49.10dangerarea[TK]D-Fender: I can understand that, but I'm trying to present ISDN to a location where there is none for ISDN data calls so I don't want to do VoIP, I actually want ISDN but using IP as a transport mechanism
19:49.41[TK]D-Fenderdangerarea: * is NOT an ISDN switch and packet-relay platform
19:49.58[TK]D-Fenderdangerarea: FORGET about "data" over this.  Completely
19:50.10[TK]D-Fenderdangerarea: * is a voice platform
19:51.04NavionI'm looking for information about using Asterisk for radio station call in. Kind of like an ACD but different.
19:51.22dangerarea[TK]D-Fender: I thought as much, but one of those if you don't ask, you never know
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20:02.29NavionBudget? lol
20:03.12QubeZi have setup roundrobin for my queue rather than rrmemory.  Is there a way to control which order it goes in?
20:04.02QubeZPenalties do not seem to work because when I setup penalties on my local members (cell phones), it only bounces between my internal office extensions and doesn't go to the local members. Basically, I'm looking to dial office phones first and after each of them (2 total) has rung, it goes to cell phones.
20:04.17p3nguinroundrobin is deprecated.
20:04.42p3nguinAnd penalties are not for that purpose anyway.
20:05.37p3nguinhttp://www.voip-info.org/wiki/view/Asterisk+call+queues
20:05.42p3nguinThat should explain the use of penalties.
20:05.48*** join/#asterisk jhirley (~jhirley@c-75-74-13-194.hsd1.fl.comcast.net)
20:06.26p3nguinIf you want to re-order the members that get called, try re-ordering them physically in the conf.
20:06.44NavionIt'd be cool to have a full blown app for this. I bet it could be commercialized or at least be a very popular item. Combined with Rivendell automation.
20:10.39NavionOh yeah, I looked at the. Rivendell is free and seems to have everything. It came out of Salem Broadcasting a Christian Broadcast group. Now there is a semi-commercial group supporting it. Runs under Linux.
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20:15.18NavionWe'll have a complete studio with CD's and Turntables. The automation will just be a coupl of inputs to the board.
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20:25.31Kattyis it 5 yet.
20:26.02dadavoipnope
20:26.06Katty:<
20:26.09dadavoiplol
20:26.41dadavoipKatty are you m or f?
20:26.52Kattydoes it matter
20:26.59dadavoipnope just asking
20:27.42Kattyk
20:27.42dadavoipsoooooo
20:29.04p3nguindadavoip: How about you?
20:29.21KavanSp3nguin, it's irc, let's be serious...chances are he's a dude! :P
20:29.39dadavoipi am of the species that i heard originated from mars
20:29.49p3nguinCould be.
20:29.50dadavoiplol
20:31.34dadavoipanyone familiar with the working of astgui, the way it interacts with asterisk?
20:31.42luke-jrany way to get Asterisk to transmit silence while recording Voicemail?
20:32.05dadavoipluke-jr-?????????
20:33.02KattyUSE THE FORCE
20:33.07luke-jr:/
20:33.15Kattyperk up buttercup
20:33.34*** join/#asterisk ahowlader (Adnan@180.234.39.137)
20:35.07luke-jrmy origination is hanging up if there's no audio data in 10 seconds
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20:37.03*** join/#asterisk generalhan (~asd@about/windows/staff/generalhan)
20:37.09dadavoipso if i understand you want to play no message when you link to voicemail
20:37.58luke-jrno
20:38.01luke-jrjust silence
20:38.04KobazAnyone familiar with something like this: http://www.n1wireless.com/Jabra_Bluetooth-Jabra_A210_Bluetooth_Adapter_for_2_5mm_headset.html
20:38.38dadavoipluke-jr: do you want the user to leave a message?
20:38.42luke-jryes
20:39.09luke-jrdadavoip: the greeting works fine
20:39.18luke-jrit's the recording that fails because Asterisk stops transmitting audio
20:39.30*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
20:39.30generalhanhey all, we just moved offices, and now after getting my asterisk server back online i seem to be having an issue. when a call comes in through our PRI into any of the SIP phones the caller cannot hear, though the called party can hear just fine. any suggestions on how to get to the bottom of this ?!
20:39.59dadavoipso if a guy is busy on phone the user will be routed voicemail have no message how is the guy going to know when to leave a message
20:40.14*** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-kyhpeiypkpcpnptv)
20:40.41luke-jrdadavoip: like I said, the greeting works fine
20:41.02dadavoipbut you dont want the greeting
20:41.33luke-jrsure I do
20:41.39*** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-ctujduapkdgzsauh)
20:42.06dadavoipsorry i am ot getting what you wanna do?
20:43.19dadavoipMaxsilence defines how long Asterisk will wait for a contiguous period of silence before terminating an incoming call to voice mail. The default value is 0, which means the silence detector is disabled and the wait time is infinite. Maxsilence takes a value of zero or a positive integer value which is the number of seconds of silence to wait before disconnecting.
20:43.21luke-jrdadavoip: transmit silence while recording the message
20:43.46generalhanstrange ... now working just fine ?!
20:44.51luke-jrdadavoip: Asterisk isn't ending the call, the caller is
20:44.59*** join/#asterisk Da-Geek (~Da-Geek@nat/redhat/x-jyflkmccbqnrcela)
20:45.58dadavoipexplain in momre detail it late, can't think with no info
20:46.25luke-jrdadavoip: when Asterisk begins recording the message, it stops sending UDP audio packets to the caller
20:46.43luke-jrGoogle Voice, the caller in this case, sees this, and gives up on the call after 10 seconds, assuming it's been lost
20:46.46dadavoipyes because its recording the message
20:47.16luke-jrI want Asterisk to keep sending audio UDP packets while it records
20:47.19luke-jrjust silence
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20:47.40[TK]D-Fenderluke-jr: It should... * doesn't support silence suppression anyway
20:47.54Qwell[TK]D-Fender: it's not surpressing anything
20:47.58dadavoipi dont think that is possible
20:48.01Qwellthere's nothing to suppress..
20:48.11luke-jr[TK]D-Fender: it doesn't generate silence when there's no audio, is the problem
20:48.25[TK]D-FenderQwell: Yes, well that normally comes with CNG attacted and a stop of RTP, etc
20:48.30tzangerif there's no audio isn't it silent by definition? :-)
20:48.32Qwellluke-jr: did you try enabling the transmit_silence option?
20:48.42luke-jrQwell: this system is still on 1.4
20:48.52[TK]D-Fendertzanger: UltraQuiet (tm) :p
20:48.56dadavoipluke-jr: you might be able to send the call to a meatme the record the meetme
20:49.02Qwellthen no, you can't.
20:49.12luke-jr☹
20:49.36luke-jrcan I get Asterisk to send a "beep" sound every 9 seconds? <.<
20:49.43dadavoipyes
20:50.10dadavoipnot while recording the message though
20:50.12luke-jr-.-
20:50.17Kobazdadavoip: sure why not?
20:50.55Kobazyou can record, beep, bark, whatever
20:51.17dadavoipif the beep is commping from the far side
20:51.54Kobazdadavoip: you're not making any sense
20:52.34dadavoipluke-jr wants to send a voice stream whiles recording a message
20:52.42Kobazthat's fine
20:52.55dadavoiphow can you send a beep every X seconds?
20:53.06dadavoipwhiles recording the call?
20:53.18KobazMixMonitor(), originate, chanspy, and some local channels
20:53.54dadavoipbut he was asking if this is a voicemail features
20:54.16Kobazor you can abuse the L option of Dial() and use a very large limit... say maxint.. and set the reminder time to 9 minutes
20:54.19QubeZim using asteriskqueuemembers table so not sure how to reorder in this
20:54.27Kobazdadavoip: no it's not built in... but you can write dialplan to do it
20:54.29QubeZi want a specific order for the calls to ring
20:54.57dadavoipthats what i say, i would push it to a conference and record the conf
20:55.19Kobazand you can use the limit options to play sounds
20:55.29dadavoipQubeZ: use ring groups
20:57.05dadavoipbut if you in a meetme you are sending voice silence, so goolgetalk would not disconnect
20:58.01dadavoipso no need to send the tones
20:58.15dadavoipany guru available to help with a project paid?
20:59.44Kobazit would be helpful to know the details of the project
21:03.39Kobazwhere's everyone going
21:03.59Kobazdid someone trip over a power cord again
21:04.05p3nguinqubez: Why didn't you do what I said an hour ago?
21:04.44*** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca)
21:05.34dadavoipi have customer that has a vicidail server 10 seats, very simple, working fine, vicidial using its own astguiclent to link to asterisk diaplan with some pearl agi, when an outbound call is made they pass the number to be dialed to the default context in asterisk
21:06.21dadavoipso that means they are sending the number to  be dialed, i need to get the agentinfo also passed to the diaplan bfore the call links to trunk
21:06.52dadavoipi created a call credit check that i wan to add to vicidial
21:07.17dadavoipso every outcall must check if the agent has crdit available
21:07.25dadavoipdoes this make sence
21:07.27dadavoip?
21:08.35Kobazyeah
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21:09.04dadavoipcan u help?
21:09.09Kobazperhaps
21:09.15QubeZp3nguin: sorry i missed it, had to jump into a conference call
21:09.27QubeZp3nguin: can you reiterate? sorry
21:10.13dadavoipwould you know how to go about it, i just started learning php?
21:11.03dadavoipQubeZ: use ring groups
21:11.31QubeZthanks will do
21:12.10Kobazdadavoip: you would have to incept the callout... and php is probably a crappy way to do it
21:12.50dadavoipvicidial uses pearl, next lang on my list to learn
21:12.58Kobazperl.. not pearl
21:13.08p3nguinqubez: Don't use ring groups.  Order the members in your queue config in the order that you want them to be rang, plus use a ring strategy that allows for them to be called individually.
21:13.42QubeZp3nguin: all of my stuff is database driven except the dialplan. Queues and members are in database.
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21:13.52dadavoipsorry i ment queues not ringgroups
21:14.07p3nguinqubez: Unless, of course, you don't need to be using queues in the first place, in which case proper dialplan should be the answer.
21:14.37QubeZdadavoip: i am using queues, i have all 6 members in the queue - 3 office phones and 3 cell phones but the cell phones never get called
21:14.55p3nguinUnder what circumstances should the cell phones be called?
21:15.20p3nguinShould they have the same "weight" as a regular in-house phone?  Should they only be called when no other in-house phones are available?  Etc.
21:15.48QubeZp3nguin: I would like it office #1, office #2, cell #1, cell #2, office #3, cell #3 --- #3 is me and should be the last in line of escalation but penalties dont appear to work the way i thought they did
21:16.10p3nguinPenalties are described in the link I gave you earlier.
21:16.21QubeZp3nguin: right exactly. I need the first two office and cell phones (my 2 helpdesk guys) to get called first - office then cell, if they do not answer (4 diff phones) then it rings me.
21:16.27dadavoipp3nguinmade me doubt my selve it is ring groups: roundrobin
21:16.32QubeZp3nguin: yup, i read on penalties and they do not apply here
21:17.11QubeZright now im using roundrobin because i dont want rrmemory. If it ends up last call on 4th phone, i dont want the next one to be my cell phone. I need it to start back again with the helpdesk guys office phones
21:17.11p3nguinPenalties only work when the first priority phones are NOT AVAILABLE and then the next priority level will be used.  If the phones in the first priority level are able to take more than one call or simply do not answer, the other priorities will NEVER be used.
21:17.32QubeZp3nguin: right, so it doesn't work for me. How else can I accomplish this?
21:17.42p3nguinCascading queues.
21:18.22QubeZis that the most direct way to do what i want?
21:18.40QubeZput myself and my boss in a separate queue and only have that queue called if the first level of techsupport is not answered?
21:18.49bougymanyes
21:18.50p3nguinYou weren't satisfied with my original suggestion of listing the members in order, so yeah, that's about all you have left.
21:19.12QubeZp3nguin: i am satisfied with that, i dont mind listing them in order but i have this database driven so it doesn't appear to work this way
21:19.20*** join/#asterisk defsdoor (~andy@plingit.gotadsl.co.uk)
21:19.31*** join/#asterisk nny (~Scott@174.107.201.103)
21:19.37p3nguinThen allow for escalation through multiple queues.
21:19.46p3nguinIt's a very easy thing to accomplish that way.
21:19.50QubeZok
21:20.21p3nguinqueue-office, queue-cell, queue-management, etc.
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21:23.32nnyi am tetsing using MixMonitor to record calls from a specific number. Normally when using Automon the file format is such that provides the time + dialed from and dialed to. Is there a convenient way to do this via the dialplan? the options for Both MixMonitor and Monitor seem to require a file name.
21:24.00nny*testing MixMonitor to automatically record calls from a specific number*
21:24.30dadavoipnny you can use simple dialplan concepts to do this
21:24.39p3nguinMixMonitor(${STRFTIME(${EPOCH},,%Y%m%d)}-${CALLERID(num)}.gsm)
21:24.57p3nguin= 20100908-3149691077.gsm
21:24.59nnyyeah figured as much, just wondering if I could use the parameters from Automon
21:25.06nnyp3nguin: thanks
21:25.07dadavoipwhat p3 said
21:25.38p3nguinYou could probably figure out the called number and add that, too.
21:26.17nnythis will suffice, thanks!
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21:29.25Get_The_Fishhello all... is there a built-in variable that has the name of the context that called a sub-routine?
21:29.44p3nguin${CONTEXT} not what you're looking for?
21:30.22Get_The_Fishno, I need the context from which the GoSub was called
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21:31.45upbi think the dialplan concept is just genious, allowing people to cram programs into ini file syntax
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21:31.51dadavoipset(var=subcontext) then you can call it from the other context i think
21:31.51Get_The_FishI know that there is one for a macro (MACRO_CONTEXT), looking for something like that
21:32.01bougymanupb: Shirley, you jest.
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21:32.27Get_The_Fishdadavoip, thanks, that is probably what I will have to do
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21:33.20dadavoipGet_The_Fish did you get the fish
21:33.40Get_The_FishI get fish like Trump gets money
21:35.38dadavoiplol
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21:44.03LetoricAnybody know why when using app_monitor, the uniqueid that is recorded in cdr-csv/Master.csv doesn't match the audio files in /var/spool/asterisk/monitor/ ?
21:44.09Letoricit's usually close, but not identical
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21:45.16dadavoiptwo dif unique ids
21:46.58Letoricok, so let me phrase this as goal oriented  ;) I would like to be able to rapidly find the recordings associated with a particular call. Currently I'm using the uniqueid and just finding the close number, which seems to work ok, but I'd like to be able to pass off the task to somebody and have them go through minimal thinking
21:47.12LetoricHow could I tell a secretary how to find the file, as an example
21:47.36dadavoipchange teh cdr variable
21:48.34LetoricNot sure I know how to do that.
21:48.56dadavoipadd another variable link it to your mix_monitor file
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22:15.07ShaunRi need an alternative to voicepulse for sip trunking since voicepulse restricts international calling.  Any recommendations?
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22:19.25lvlolvlothere are many providers out there you can try http://www.voip-info.org for a particular reference
22:19.31lvlolvloI personally use Flowroute
22:20.06p3nguin~trunk
22:20.06infoboti heard trunk is a word with varying definitions.  In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN.  There is no such thing as a "SIP Trunk" -- Don't use the term. the nose of an elephant
22:21.26WIMPyWouldn't it make sense to use an itsp in the country you're calling to?
22:22.05lvlolvlono, i would think it would make more sense to use an itsp in the country your box is in
22:22.16p3nguinYeah it would.
22:22.30p3nguinLess latency in the VoIP end of things.
22:22.32lvlolvloas it would term over coper and not be voip with latency
22:22.43p3nguinLet the ITSP deal with the landline crap.
22:22.44lvlolvlocopper(
22:23.00lvlolvlolol
22:23.02WIMPyWhat's the point of uing voip then?
22:23.24lvlolvlocost, capacity, overhead...
22:23.29lvlolvlothere are many
22:23.33lvlolvloi would imagine
22:23.45QwellYou'd get far lower cost using a provider in the location you're calling to/from.
22:23.50p3nguinI use VoIP for two reasons: cost and convenience.
22:24.11lvlolvlobut somethings are just better suited for non-voip (e.g. faxing reliability)
22:24.19lvlolvloi agree with p3nguin
22:24.29lvlolvloconvenience is my driving factor
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22:25.13ShaunRflowroute is the one i was looking for thanks
22:25.35WIMPyConvenience is more like the distractiong part.
22:25.41ShaunRi push domestic calls out my t1, but international is waaaay cheaper over a sip trunk
22:26.08p3nguinEven though there is no such thing, you insist you're using it?
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22:26.41lvlolvlomagic!
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22:31.49citywokin realtime * do you have to have a declaration for every context, and then switch => it, or is there a way to have a global type switch
22:32.36[TK]D-Fendercitywok: Former.
22:32.59citywokhmmm. that's annoying lol.  sort of defeats my use
22:33.36citywoki'm creating the contexts on the fly, so having to create them in extensions.conf to siwtch them to avoid the reload... doesnt work. i still have to reload.
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22:49.19Belgarath-/j #vps
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23:34.34lost_sou1wondering..  I know it's not "best practice" to run alot of services on a router system.  However I'm wondering if their are any critical bugs which would make running asterisk on my router a terrible idea?
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23:42.42jimi_When I am in the asterisk cli, how can i get debug/output only related to a specific dialplan context... or a specific peer, etc
23:45.18lost_sou1set sip debug peer then the ip maybe
23:46.11jimi_lost_sou1, ok, ty... trying to do call tracing, find out why calls are getting stuck in queues without people answering them, etc
23:47.07[TK]D-Fenderthat won't help, and no, you can't
23:47.43lost_sou1[TK]D-Fender: ty for the correction
23:49.39jimi_What's the method for diagnosing these?
23:51.14[TK]D-Fenderjimi_: Large scroll-back buffers and lots of staring
23:51.35jimi_[TK]D-Fender, ok, i thought there might be tracing methods.
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23:54.13[TK]D-Fender[19:47]<[TK]D-Fender>that won't help, and no, you can't <- you seemed to acknowledge this... and then backtracked
23:56.04jimi_when did i back track?
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