IRC log for #asterisk on 20081018

00:00.09*** join/#asterisk CodeWarrior_ (n=kleber@201-74-223-165-am.cpe.vivax.com.br)
00:00.14ShamusNYhi guys
00:00.36ShamusNYhas anyone tried recording a conversation in wireshark?
00:00.49ShamusNYi have 99% of one, but i am missing the INVITE
00:00.49hardwirelike.. can't he jsut remove waitexten?
00:00.54hardwirethen loop?
00:00.57sdanielsAhhhhh, i think i understand.... so just because someone is in a context doesnt mean that that is the only place it will try to match the regex, it goes though all contexts that are included?
00:01.05ShamusNYso wireshark won't decode the RTP/UDP stream
00:01.07ShamusNYany ideas?
00:01.15ShamusNYor suggestions for a channel that might help?
00:01.15hardwiresorry.. I know.. wrong tree right.
00:01.18CodeWarrior_hello folks, I'm totally newbie about Asterisk, but I have many doubts about how to starting, is there any good soul that can introduce me ?
00:01.20hardwireboo
00:01.33lesouvage~book
00:01.34jbotmethinks book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
00:01.42*** join/#asterisk ganeshix (n=chatzill@cpe-24-29-46-121.nycap.res.rr.com)
00:01.56hardwiresdaniels: add ||ivr after m
00:01.57lesouvageCodeWarrior: welcome to the crowd
00:01.57hardwireinn Background
00:02.07aiksa[LV]sdaniels: pretty close
00:02.17CodeWarrior_lesouvage: hehehe
00:02.32aiksa[LV]it will wait to see if you wanted to "go further"
00:02.54ManxPowersdaniels: Your current dialplan would allow anyone that could get to the IVR to dialout your lines
00:02.56aiksa[LV]lets say you extensions 1,2,200,300,4
00:03.03ManxPowerthis.  is.  bad.
00:03.11aiksa[LV]ManxPower: you spoild the triumph
00:03.13*** join/#asterisk boolean12 (n=boolean1@tandem.uplinktel.com)
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00:03.20*** mode/#asterisk [+o Corydon76-dig] by ChanServ
00:03.20lesouvageCodewarrior: google on "asterisk guru" and you will be pointed to a website with info that helps me a lot.
00:03.22[TK]D-Fendersdaniels: No
00:03.27aiksa[LV]Manx not neceseraly
00:03.33aiksa[LV]he includes ivr in phones
00:04.03aiksa[LV]if he included only ivr in some incomming context
00:04.06sdanielsadding the ||ivr made it work.
00:04.09boolean12Does anyone know a good way to list all the realtime config families?
00:04.13[TK]D-Fendersdaniels: because you aren't in the [ivr] context.  You are in [phones], from a functional standpoint imagine that as being a cope& paste of everything included into ONE mess.
00:04.18hardwiresdaniels: but instead of just doing that
00:04.19*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
00:04.19*** mode/#asterisk [+o mog] by ChanServ
00:04.28hardwireyou should put extension 900 in phones
00:04.37hardwireand have it "goto" ivr,900,1
00:04.44CodeWarrior_lesouvage: I'll friend, but at this moment, let me start asking you about something: Do I need to have some extra hardware to try Asterisk, or just install it, setup it and voila, it's working ?
00:05.06hardwiresdaniels: any time you allow a user to dial an extension, you should isolate it as much as possible
00:05.07ManxPowerhttp://www.fnords.org/~eric/dialplan-example-v2.txt
00:05.11[TK]D-Fendersdaniels: In fact your ivr should have been run off the "s" exten and not "900".  In your IRV you can dial 900 over and over to loop indefinitely
00:05.13ManxPowerthat's off the top of my head.
00:05.23aiksa[LV]and the same later on in [incomming] context to make it public teh safe way
00:05.38hardwire[TK]D-Fender: thats a good tip
00:05.41ShamusNYanyone?.
00:05.44aiksa[LV]tzafrir_laptop: ping again.
00:06.33sdaniels[TK]D-Fender: I think I get it...
00:06.51ManxPowersdaniels: contexts (and this is what you have -- a context problem) are one of the hardest things to understand in Asterisk and maybe the most CRITICAL thing to understand.  Contexts are the security mechanism of Asterisk.
00:07.12sdaniels[TK]D-Fender: Im only about a week or so into Asterisk. thanks for the info yall.
00:07.13hardwireaiksa[LV]: does xorcom have a support line?
00:07.24aiksa[LV]no just email
00:07.32hardwireare they gone for the day?
00:07.37LiNeTuX_Homeaiksa: I know Xorcom.  I might be able to help.
00:07.50ManxPoweraiksa[LV]: naw, just holler tzanger!!! and he eventually materializes.
00:07.59sdanielsIm going to try and make this config right, I may pop back in to get a proof read of my config if yall dont mind. Thanks again for the info.
00:08.01aiksa[LV]:)))
00:08.06[TK]D-FenderManxPower: you mean tzafrir_laptop IIRC...
00:08.15[TK]D-Fendersdaniels: np
00:08.17ManxPoweryeah, him too!
00:08.25aiksa[LV]LiNeTuX_Home: ok lets see if you can help
00:08.51aiksa[LV]Distrib - Slack.
00:08.57aiksa[LV]2.6.25.18-smp kernel
00:09.13aiksa[LV]trying to add Astribank BRI (8 ports)
00:09.40aiksa[LV]the bank works just fine from demo CD (so - no HW error).
00:10.06[TK]D-Fendersdaniels: What the hell, I'm feeling generous : http://pastebin.ca/1229815
00:10.14aiksa[LV]As per Xorcom manual - I have built their distribution of bristuff
00:10.25*** join/#asterisk InHisName (n=InHisNam@c-71-225-221-149.hsd1.pa.comcast.net)
00:10.49aiksa[LV]now - as far as i can see from lsusb and zaptel_hardware
00:11.08aiksa[LV]the chan.bank itself has been identified and even reflashed
00:11.35*** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com)
00:11.42devilsoulblackhi
00:11.45aiksa[LV]./zaptel_hardware returns "usb:001/002          xpp_usb+     e4e4:1142 Astribank-BRI FPGA-firmware
00:12.16aiksa[LV]the point is - i dont see the actual isdn2 spans
00:12.26CodeWarrior_Do I need to have some extra hardware to try Asterisk, or just install it, setup it and voila, it's working ?
00:12.47[TK]D-FenderCodeWarrior_: No special hardware, just a PC running *NIX
00:13.01aiksa[LV]dmesg is full of NOTICE-xpp: XBUS-00: copy_pcm_tospan: non-existing address (00): RECEIVE PCM
00:13.15CodeWarrior_but how can I make a call using asterisk ? trough Skype ?
00:13.18jayteeCodeWarrior_, a local network and a workstation you can run a softphone on at minimum
00:13.28aiksa[LV]and under /proc/xpp/XBUS-00 there aind XDB subfolder
00:13.30jayteeSkype no!
00:13.38aiksa[LV]LiNeTuX_Home: - any idea?
00:13.53aiksa[LV]CodeWarrior_: not yet
00:13.54jaytee~softphones
00:14.05jaytee~softphone
00:14.06jbot[~softphone] A soft-phone is a program that lets your computer use a compatible audio interface (usually a speaker&mic, or headset), and optionally a webcam, to place VoIP calls with a given protocol (SIP/IAX/H.323/MGCP/etc). For links enter ~zoiper , ~xlite , ~twinkle , ~bria , ~eyebeam , ~ekiga
00:14.06CodeWarrior_ok, no skype, but what other company can provide me the "phone line" ?
00:14.22aiksa[LV]CodeWarrior_: plenty of them
00:14.31LiNeTuX_Homeaiksa[LV]: thinking... checking against my wiki notes...
00:14.56hardwireaiksa[LV]: whats /etc/zaptel.conf look like?
00:15.12ManxPower~itsp
00:15.13jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
00:15.30CodeWarrior_ah ok, so I have to had some company to provide me the VoIP line, right ?
00:15.43jayteecorrect
00:15.48hardwireCodeWarrior_: some company has PSTN access.. and trunks that to VoIP signals and RTP
00:15.55hardwirefor joo
00:16.10CodeWarrior_ok, what are that E1 interfaces and so on ?
00:16.29hardwireCodeWarrior_: legacy inventions left over from egyptian pyramid times.
00:16.34ManxPowerCodeWarrior_: now is the time for you to go read The Book
00:16.36ManxPower~book
00:16.36jbothmm... book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
00:16.49jaytee~101
00:16.50jbotwell, 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony.  You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf
00:17.04ManxPowerjaytee: Your jbot-fu is strong.
00:17.12hardwiresdaniels: you make it sexy yet?
00:17.14CodeWarrior_great ! I'll get the both books
00:17.15jayteeManxPower, thanks.
00:17.30LiNeTuX_Homeaiksa[LV]: do you have /proc/xpp/XBUS-00/summary
00:17.46[TK]D-Fender~e1
00:17.47jbot[~E1] E1 is the basic digital telephony circuit used everywhere except the US,Japan,Taiwan & Hong Kong (who use T1 and similar variants). E1 runs at 2.048Mbps. It can be an unstructured channel for data. It can be channelized, to provide 31 time slots of voice or data, each of 64kbps. Time slot 16 is used for D-Chan when used with PRI signalling.
00:17.51jayteeCodeWarrior_, start with the Asterisk book, use the other one for unfamiliar terms you may come across
00:18.05unpaidbillhuh, chan_ooh323 appears to be a pain in the ass
00:18.23unpaidbillwont work with t38modem properly
00:18.25hardwireaiksa[LV]: you have an /etc/zaptel.conf.. right?
00:18.26jayteemajor pain and not really worth it
00:18.52hardwireunpaidbill: you dropped the t38 bomb?
00:19.04CodeWarrior_jaytee, ok I will, thanks !
00:19.10*** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com)
00:20.09CodeWarrior_another question, just having a computer + asterisk and a VoIP provider, can I have an ACD ?
00:20.30jayteeCodeWarrior_, ACD as in queues? Yes
00:20.32hardwiresure
00:20.37hardwireruns away
00:20.45unpaidbilloops
00:20.46unpaidbilldid i say that
00:20.50CodeWarrior_very nice
00:21.13unpaidbilli meant t triple 8
00:21.25unpaidbillmy name is john connor and the only way to communicate with the future is through h323
00:22.10jayteeunpaidbill, if you ever get that comm link working make sure to tell them to cut the red wire, not the blue one.
00:22.48CodeWarrior_another question, how could I have telephones to use as ramals connected to asterisk ? special hardware ?
00:22.54CodeWarrior_ATA ?
00:23.36jayteewhat's a ramal?
00:24.15unpaidbillperhaps he meant jamals
00:25.14aiksa[LV]LiNeTuX_Home and hardwire sorry just got back. remote desktop session terminated out of the blue
00:25.15CodeWarrior_internal lines connected to pbx system
00:25.24CodeWarrior_to transfer calls
00:25.51aiksa[LV]hardwire: yes i have it of course
00:25.57jayteeinternal lines? you mean analog phones as internal "extensions"?
00:26.00[TK]D-FenderCodeWarrior_: To use an analog phone with * : Linksys PAP2T-NA for example
00:26.23aiksa[LV]hardwire: if i add even a single span, ztcfg fails that it cannot initialize the device
00:26.25CodeWarrior_ah right
00:26.27[TK]D-FenderCodeWarrior_: to use LINES, usually we use specialized PCI cards, or sometimes similar gatways
00:26.45CodeWarrior_or voip phones, right ?
00:27.10aiksa[LV]LiNeTuX_Home: http://www.pastebin.ca/1229821
00:27.24jayteeCodeWarrior_, the book explains FXO (for analog lines from your telco provider) and FXS for analog phones.
00:27.42CodeWarrior_great
00:28.07CodeWarrior_so, I think the basics are answered, right now I need to go to the books
00:28.34CodeWarrior_thanks a lot guys, I hope soon I can have an Asterisk running fine ;)
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00:29.22aiksa[LV]LiNeTuX_Home: thats the output of XBUS-00/summary
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00:35.14dennisharrisonanyone know what fd == -1 in astman_append, should not happen means ?
00:38.35jayteedennisharrison, ask in asterisk-gui, it's an asterisk-gui specific error:  http://lists.digium.com/pipermail/asterisk-gui/2007-September/000745.html
00:38.56dennisharrisonjaytee, howdy, asterisk-gui is dead right now :(
00:38.58dennisharrisonoh well :)
00:39.00dennisharrisonfreepbx it is!
00:39.28aiksa[LV]_afkok. thats it; time to bed.
00:39.45dennisharrisongoodnight
00:40.02jayteeit's bedtime in latvia but not here
00:41.32jayteewonders if glucosamine-chondrotin actually works or if it's just a bunch of hype cuz he's starting to sound like a bowl of Rice Krispies right after you add the milk.
00:41.46aiksa[LV]_afkjaytee: i was just stating the fact, that its the high time to go to bet for me
00:42.13jayteedon't forget to say your prayers
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00:42.29aiksa[LV]_afkoh no
00:42.33aiksa[LV]_afknot yet :)
00:42.38aiksa[LV]_afkLiNeTuX: alive?
00:42.56LiNeTuXaiksa[LV]: sorry, battery died!  Gotta give my daughter a bath... bbl...
00:43.30aiksa[LV]_afkok. did you see the pastebin result?
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00:46.24aiksa[LV]_afkok, LiNeTuX take care. I am off for a sleep now
00:46.37aiksa[LV]_afk4hrs of sleep left for me:P
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00:56.35CodeWarrior_bye folks, again, thanks for the answers... see you..
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01:34.11Arnavhi.. my service provider just switched to freeswitch and my calls in asterisk don't work anymore, please help me...
01:35.32Arnavis the freeswitch better than asterisk???
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02:12.37jayteeit's quiet...............too quiet
02:13.05[TK]D-Fendergoes to hide the last of the bodies
02:13.18jblackThat reminds me.... /me checks on dance of the dead
02:13.26jayteehands [TK]D-Fender a bag of lime to help speed the decay
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02:27.34coppicereducing the time constant will speed the decay
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02:29.19jayteeso you advocate something that could potentially destabilize the space-time continuum?
02:36.01coppiceno, I'm not at Cern
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02:44.13jblackIt's no fair. Why do the theoretical physicists get to end the world?
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02:50.34jayteeit won't be the LHC that ends the world. People forget about the "Butterfly Effect" or how a butterfly flapping it's wings on one side of the world kickstarts a chain reaction that creates a hurricane on the other side of the world. It'll start with the simplest of things, a delivery truck overturns on the freeway and the beer it is carrying never arrives at it's destination, a VFW Hall in New Jersey. From that little event things will start to u
02:50.34jayteenravel.
02:54.52coppicejaytee Butterfly Effect was a terrible movie
02:55.44jayteecoppice, I was talking about the concept, not the movie.
02:55.58coppiceduh!
02:56.22jayteeand yeah, the movie kinda sucked regardless of which ending you watched
02:57.56coppicekatrina probably started with a burrito. the owner farted, and step by step the wind increased
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02:59.20jayteeanything's possible :-)
02:59.27WimpManSo that's the master plan wor WW3? Free Burritos for everyone?
02:59.45coppiceBurritos - the wind of change
03:04.32denonhaha
03:04.34denonthat's just wrong coppice
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03:06.53jblackimagines what a burito made by CERN would be like.
03:07.34WimpManMore like a Donut.
03:07.37jblackI suppose it would have 11 layers, require liquid nitrogen, and be so massive that it would swallow the earth up in a black hole.
03:08.10coppicewell donuts started out straight in china, so maybe a ring shaped burrito is the next step
03:09.22WimpMandid indeed wonder if a ring Burrito would be a good idea.
03:09.46WimpManBut I fear it wouldn't be fast food any more then.
03:10.42jayteehttp://beyondrandom.com/wp-content/gallery/randompics-august/tape.jpg
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03:10.54jblackdance of the dead is great.
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03:25.17Kate-oHey
03:26.43Nuggetjaytee: http://macnugget.org/photos/tboicey_comics/welders
03:27.36jayteehehe
03:28.21mog<PROTECTED>
03:28.28Qwellhuh?
03:28.37Qwelld'oh
03:29.03jayteejabber collision?
03:29.06Qwellthat better?
03:29.11moghopefully
03:29.26Qwellsorry, just brought my desktop back up..  not sure how laptop had /Home
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03:30.23lolipopsI just installed asterisk 1.6 on my ubuntu box. /var/run/asterisk.ctl doesn't get created anymore. other than that, everything works fine. any pointers?
03:30.39Qwelllolipops: installed from a package?
03:30.47lolipopsQwell, no, from source.
03:30.56Qwellthen that file should exist
03:31.05jblackon ubuntu, it should be /var/run/asterisk/...
03:31.12Qwelljblack: hence the question above.
03:31.21lolipopsits not under /var/run/asterisk either.
03:31.31lolipopsmy understanding was that the socket was created at launch?
03:31.37Qwellit is
03:31.52lolipopswell it seems that it's not happening right now... :/
03:32.01lolipopsi looked under both /var/run and /var/run/asterisk.
03:32.14Qwellhave you verified that it's 1.6 running, and not some odd mix-n-match?
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03:32.42lolipopsasterisk -V returns Asterisk 1.6.0.1
03:33.00Qwellthat shows the version of the client you just launched
03:38.03[TK]D-Fenderlolipops: How did you start *?  can you confirm that its running?
03:38.38lolipops[TK]D-Fender, i started with some /etc/init.d script. it is running; i can place and receive calls.
03:39.32[TK]D-Fenderlolipops: I'd verify the script you used.
03:39.58lolipops[TK]D-Fender, it worked as advertised with 1.4; should it make a difference?
03:40.51[TK]D-Fenderlolipops: Try stopping * and starting it as a daemon yourself
03:41.30lolipops.. apparently it works as advertised as root, too.
03:41.36lolipopsi guess its a permission issue or something..
03:42.04[TK]D-Fenderlolipops: Quite likely
03:43.09lolipopsis there an easy way to tell asterisk to create the file somewhere else?
03:44.27[TK]D-Fenderlolipops: you'd have to look inside your init scrip
03:44.43lolipopsi see. well that makes sense. thanks.
03:44.58lolipopsto you too, Qwell.
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03:58.22ShamusNYI tried recording a conversation in WireShark/Ethereal.  Unfortunately, I have 99% of a the packets from a  personal call (that I need to to hear) but, I am missing the INVITE message.  The symptom is that Wireshark cannot "discover" the RTP/UDP Stream!   Do you have any ideas or suggestions to solve this?
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04:28.20hescohow much should I reload after editing voicemail.conf?
04:28.33lolipops[TK]D-Fender, I figured my problem.
04:29.29lolipops[TK]D-Fender, the /etc/asterisk/asterisk.conf that ships with ubuntu hardy is broken; it defines directories under [global] instead of [directories]
04:30.34[TK]D-Fenderlolipops: And earlier you said you installed from source...
04:30.55lolipops[TK]D-Fender, over an old 1.4 install from ubuntu
04:33.03*** join/#asterisk drmessano (n=nonya@pdpc/supporter/active/drmessano)
04:33.47hescoI'm trying to register this server to a DID provider.  Can anyone advise how I can inspect whether I've done that successfully from the console?  Is there some show command which might help me here?
04:34.36hescoiax2 show registry, I guess might be what I'm looking for
05:01.23drmessanohmm
05:08.02hescoI'm pretty sure in my tests of a new inbound DID number, that I'm reaching the right server with my call.  But the voicemail seems to be broken on the sample.  When I tried to use extension 1234, it left me with dead air, on which I finally hung up.  But then the console seems to have documented the interaction.  What are the most obvious places to look for issues with the voice mail?
05:08.33*** join/#asterisk SQLDarkly (n=nospam@p4-66.dsl.ecentral.com)
05:09.08[TK]D-Fenderhesco: Look at whats actually going on in CLI
05:10.06*** join/#asterisk BBHoss (n=bbhoss@c-68-62-170-33.hsd1.al.comcast.net)
05:10.15BBHoss~book
05:10.16jbotextra, extra, read all about it, book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
05:10.59murdock_utwaiting for 3rd edition now
05:13.43drmessanohmm
05:14.10drmessanoI keep getting a 100% hung system on 1.6.0SVN
05:20.11drmessanoQuestion:
05:21.27[TK]D-Fenderdrmessano: yes.... that's "well hung" ;)
05:21.43drmessano<SIP/111-092a3908>AGI Rx << VERBOSE "ExtensionState: 0" 4
05:21.44drmessano<PROTECTED>
05:22.03drmessanoAfter a LOOOOOOONG pause when dialing a call, thats the next thing I get on the CLI doing a debug
05:22.11drmessanoThat appears to be what was being "waited on"
05:22.31drmessanoAfter 12-15 seconds
05:23.02drmessanoHappens with any extension..
05:23.38drmessanoAny thoughts on "ExtensionState" lookup?
05:24.25[TK]D-Fenderdrmessano: Look at the full debug, and examine where that occurs in the AGI
05:24.59drmessanowhere that occurs in the AGI <-- in the dialparties.agi itself?
05:25.09hesco~paste
05:25.10jbotwell, paste is http://rafb.net/paste/, or see also pb
05:25.55drmessano~pb is better
05:25.56jbot...but pb is already something else...
05:25.59drmessano~pb
05:26.00jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
05:28.00hescohttp://rafb.net/p/s9M07H70.html <-- I'm commenting out those out of freq sound files until I can get those fixed and trying again.
05:28.09*** join/#asterisk DonX (i=don@another.lostserver.net)
05:28.12SQLDarklyI have 8 servers. 2 openSER 6 Asterisk 1.4.22(but considering 1.6) 1 ser server is redundant as well as 3 redundant asterisk boxes with QoS being handled by a cisco call manager. I am trying to find out a way to balance meetme across this cluster. Any ideas how I can acomplish this without a massive loading of modules or third party apps.
05:28.47DonXHi all...Is it possible to play a sound (.wav file) over an established call.
05:29.05SQLDarklyBoxes are also using linux ha with nic bonding. So this configuration must have room for a disaster recovery plan
05:29.25[TK]D-Fenderhesco: SCrew the console, use a soft-phone
05:29.46SQLDarklyServers are HP Proliant DL360 g4p with SLES
05:29.57[TK]D-FenderDonX: Triggered how?
05:30.07DonXcommand line maybe?
05:30.11DonXI'm not sure
05:30.50DonXI know that I can drop a call into /var/spool/asterisk/... but havent found any way to play over an existing chan
05:31.03SQLDarklyits my test lab right now so its not in any sort of production at the moment.
05:31.51SQLDarklythings can break as I have version control handled... So If anyone has any ideas on meetme clustering / balancing I would appreciate you rinput
05:32.09[TK]D-FenderDonX: I was asking what would cause this to happen in your situation...
05:32.15DonXohhh
05:32.16DonXsorry
05:32.46DonXIt would be nice to send the equiv of a wall message to everyone on my system
05:34.09[TK]D-FenderDonX: huh?
05:34.38DonXOn a unix box you can send a "wall" message to all active users on your system
05:34.50[TK]D-FenderDonX: And what does this magic term of yours mean?
05:34.56DonXit would be nice to be able to playa recording over all established calls in asterisk
05:35.12hescook, here is one which isolates the vm issue, I think: http://rafb.net/p/MKedJU13.html
05:35.18[TK]D-FenderDonX: Page + Chanspy w/ Whisper
05:35.27DonXokay thanks
05:35.27DonX:)
05:35.31DonXI'll look into those
05:36.12[TK]D-Fenderhesco: I see no call to voicemail in there
05:36.36[TK]D-Fenderhesco: And plenty of signs to stop using CHAN_OSS
05:36.46[TK]D-Fender(console)
05:38.17hescoI assume I don't need chan_oss, then.  How would I turn it off?
05:38.17WimpManHmm. The idea of a "v-wall" looks interesting. Might be nice if e.g. you run out of power. But I'm not sure if I understand the page thing.
05:39.11hescousing the sample config supplied, I asked for extension 1234, which I though was suppose to transfer to voicemail.  Did I miss something here?
05:40.52hescoAnd can you say more about what this call on console means?  I see the logs there.  I understand its a command line to try things out.  I get the sense folks are interacting directly with calls there as well.  What is that about?
05:42.04[TK]D-Fenderhesco: Stop calling console/dsp
05:42.08hescook, I did no_load in modules.conf, reloading server now
05:42.10[TK]D-Fenderhesco: And set up a soft-phone
05:42.36[TK]D-Fenderhesco: Sample configs are useless trash.
05:42.43hescoThis server is across town, not here.  I access it by ssh.
05:43.07[TK]D-Fenderhesco: So?
05:43.20SQLDarklyAlso has anyone noticed when using realtime sip they do not show in the CLI but still register? Is this an asterisk bug or am I misssing something
05:43.55hescoso you are suggesting that I set up a softphone on my desktop, that somehow interacts with this server across town?
05:49.37WimpMan[TK]D-Fender: Could you give me a hint on how Page would be usefull for that wall-type thing?
05:49.46hesco[TK]D-Fender: can you pls point me to documentation for setting up this soft-phone you advise?  I have ekiga and kphone installed locally.  How do I get them registered with the server so they get the incoming calls?
05:50.50[TK]D-FenderWimpMan: spawn a script that will call page targeting every active channel.
05:51.16[TK]D-Fenderhesco: setting up a basic SIP device is * 101.  Go look at any of the million guides out there.
05:51.44[TK]D-Fenderhesco: And they would get the incoming calls when you go and tell your dialplan to call them.
05:51.47WimpManYou can page active channels? /me takes a closer look...
05:52.28[TK]D-FenderWimpMan: page multiple local channels that will chanspy each non-local one in progress
05:53.12WimpMan*ding* Yes. That makes sense.
05:58.14*** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-207-169.phlapa.east.verizon.net)
06:17.25[TK]D-Fendercheckout time.  Later all
06:21.00drmessanoI guess I am going back to 1.4
06:21.03drmessanoThis is bullshit
06:23.18ShamusNYI tried recording a conversation in WireShark/Ethereal.  Unfortunately, I have 99% of a the packets from a  personal call (that I need to to hear) but, I am missing the INVITE message.  The symptom is that Wireshark cannot "discover" the RTP/UDP Stream!   Do you have any ideas or suggestions to solve this?
06:25.22drmessanoUse the built in recording facilites next time and not wireshark?
06:32.45*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
06:45.00*** join/#asterisk jicksta_ (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
06:46.53*** join/#asterisk wiseguy_ (n=chivilis@infospalvos.lt)
06:47.17wiseguy_hello
06:47.30wiseguy_anybody using cisco bri with zaphfc?
06:48.29Maliutano, but i think I use actual crisco oil to deep fry :)
06:48.40wiseguy_:D
06:48.49wiseguy_don't use oil
06:48.53wiseguy_use gasoline :)
06:52.14Maliutayou mean gasolene
06:52.45Maliutalike you mean aluminium
06:53.08Maliutapeople really need to learn to speak properer english
06:53.47wiseguy_i don't speak english
06:53.51wiseguy_i use globish
06:53.54wiseguy_;-)
06:54.34Maliutawhat you speak trollish?
06:54.45coppicethe proper English word is petrol
06:55.54Maliutapetroleum
06:56.00Maliutadistillate
06:56.22Maliutaso ... a troll and a flamer meet on a forum
06:56.30Maliutait'd be funny but is true
06:57.03Maliutacoppice: and in proper english there is the term gasolene
06:58.14wiseguy_;)
06:59.12coppicecoppice: in England the term gasoline is recognised, but only as a foreign word
06:59.56coppicethough I believe in the 19th century the term gasoline was used for lighting fuel
07:00.10Maliutacoppice: yeah, a fuel for lamps
07:00.18Maliutaas it was/is here in .au
07:00.41coppicebut its spelled gasoline, not gasolene
07:01.00Maliutacoppice: I think you'll find it's gasolene
07:01.20coppiceI think I just checked a dictionary to avoid looking foolish :-)
07:01.24Maliutait's how I learnt to spell it and what all my dictionaries say
07:01.42Maliutacoppice: an english dictionary or an american one?
07:01.44*** join/#asterisk sudhir492 (n=sudhir@adsl-154-183-78.mco.bellsouth.net)
07:01.49sudhir492Hi all
07:01.59coppiceenglish. I don't speak american
07:02.04Maliutais not high
07:02.18Maliutaum-er-ick-an
07:02.50Maliutaas it was famously done in a comedy sketch here (done by a yank who had his tv career in .au)
07:02.55sudhir492I dont speak um-er-ick-an either. Just know a few words :-)
07:03.08coppicethough the 16 volume OED has many spelling for many words, and will probably allow almost any misspelling you try :-)
07:03.12sudhir492only write american
07:03.18Maliutacoppice: hardda taulk um-er-ick-an
07:03.50sudhir492Anyone using asterisk queues here?
07:04.49wiseguy_sudhir492: yes
07:04.50wiseguy_;-)
07:06.50sudhir492There are three SIP phones which are shared by 8 agents. Is there a way that an agent logs in from an extension with his agent id, and becomes part of the queue and the reports will show which agent number took calls
07:06.57Maliutawe are all queued up for our turn on asterisk
07:07.09ShamusNYdo you know of another app that can use pcap and output the conversaion?
07:07.18ShamusNY-> drmessano
07:07.38wiseguy_sudhir492: yes, it is possible
07:07.40MaliutaShamusNY: why do you want to do it with pcap?
07:07.51sudhir492wiseguy_, how
07:07.54MaliutaShamusNY: this is what the inbuilt recording features are for
07:07.54wiseguy_sudhir492: just create agents, password
07:08.06drmessanoUse the built in recording function
07:08.06wiseguy_sudhir492: make them members of queue
07:08.08drmessanoI said that
07:08.12sudhir492ok, in agents.conf
07:08.18MaliutaShamusNY: unless you are trying to tap other people conversations
07:08.24wiseguy_wiseguy_: and make login in extensions.conf
07:08.26ShamusNYno
07:08.29ShamusNYi have vonage
07:08.37drmessanoROFL
07:08.39ShamusNYso i piped the vonage to my 2nd nic
07:08.56ShamusNYand recorded the traffic on the NIC to pcap
07:08.57wiseguy_sudhir492: for statistics I use queuemetrics, I think it will be enough for you
07:09.02MaliutaShamusNY: if you have a * box terminating the sip (or in the chain with a noreinvite) use * to record it
07:09.03drmessanoWelcome to #asterisk, the home of Vonage support
07:09.13wiseguy_;-)
07:09.15sudhir492wiseguy_, for login, use agentLogin?
07:09.16Maliutavonwho?
07:09.21ShamusNYwell it's just VoIP
07:09.22drmessanoExactly
07:09.26*** join/#asterisk axisys (n=axisys@117.18.230.49)
07:09.27ShamusNYg721
07:09.43ShamusNYif anyone is wellversed, this is a good time to show off
07:09.53ShamusNYcan I forge a packet for INVITE into the stream>?
07:09.58drmessanoLOL
07:10.08ShamusNYyou arent going to help
07:10.10ShamusNYthanks
07:10.26wiseguy_sudhir492: exten => 1001,1,AgentCallbackLogin(||${CALLERID(num)}@sip)
07:10.33drmessanoNo, this is a good time for us who know real VoIP to laugh out loud at someone try to con some help recording Vonage calls
07:10.51sudhir492But I saw that AgentCallbackLogin is deprecated
07:11.03MaliutaShamusNY: so this has what to do with *?
07:11.18drmessanoMaliuta: It's all VoIP, yanno
07:11.34drmessanoMaliuta: If we knew anything about VoIP, we would show off and help him
07:11.48drmessanoTHAT doesn't sound self serving
07:11.49ShamusNYit's a voip RTP/UDP conversation recorded via PCAP
07:11.52ShamusNYthats all it is
07:11.59ShamusNYyea i was baiting you
07:12.03ShamusNYbut you were being condescending
07:12.04sudhir492wiseguy_, also when the agent calls out from that extension, report will not have a record of his agent id.
07:12.08drmessanoOh, even better
07:12.42ShamusNYhow about this: have you ever used pcap to record/diagnose a VoIP call?
07:12.46drmessanoI wasn't being condescending.. that implies some level of passive aggression.  I was being outright insulting.
07:13.02wiseguy_sudhir492: for outbound calls I use queuemetrics make "virtual" outbound queue
07:13.07ShamusNYyou're an ass, and I've seen tons of you over the years in expert channels
07:13.10ShamusNYthanks
07:13.16drmessanoYou're welcome
07:13.36drmessanoTry Vonage support.. 1-VonageHelp
07:13.55wiseguy_does anyone uses cisco router with zaphfc ?
07:14.06ShamusNYis there a vonage channel
07:14.18ShamusNYi would do fine if you just helped me find the right direction
07:14.31ShamusNYthis is more about VoIP and SIP in general
07:14.32drmessano#google
07:14.44WimpManwiseguy_: What's the relationship between cisco and zaphfc?
07:14.59wiseguy_WimpMan: isdn bri
07:15.51WimpManwiseguy_: Ok, but I guess you're not trying to install * on a cisco, are you?
07:16.34wiseguy_WimpMan: no, really not. I want to peer cisco isdn interface with asterisk server with hfc isdn card
07:17.36WimpManTalking what? Voice? ppp?
07:17.46wiseguy_voice
07:18.30sudhir492wiseguy_, thanks for your suggestion. I will look into queuemetrics.
07:18.52sudhir492However, what to do about deprecated function. Is there any other way?
07:19.34WimpManOk, unfortunaletly my crystall ball is away for service. So do you try to connect them locally or what?
07:19.59wiseguy_sudhir492: it should be alternative, check voip-info.org and I am sure you will find solution
07:20.18wiseguy_WimpMan: yes, locally, via ISDN BRI
07:21.03WimpManOk, so now we know the setup. And what's the trouble?
07:21.17wiseguy_L1 (physical layer) is active
07:21.23wiseguy_but L2 and L3 fails
07:21.36wiseguy_cisco is nt side
07:21.43sudhir492thats what I am checking right now
07:22.29WimpManDo they talk the same protocoll? Any debug output from either side?
07:23.24WimpManIs it only me or does Cisco in NT mode sound scary to others as well?
07:24.46wiseguy_:-)
07:25.10wiseguy_hfc isdn card is passive, it doesnt sound good to make it NT
07:25.19wiseguy_yes, WimpMan, i have debug information
07:29.51WimpManLooks like the right time to get back to bed.
07:33.54*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
07:38.51*** join/#asterisk Samrat (i=be348cee@gateway/web/ajax/mibbit.com/x-403ad68ee3403033)
07:44.58cvnethi
07:45.07cvnethas anyone here installed a2billing ?
07:55.39drmessanoTHE DEVIL HAS
07:55.40drmessanook no
07:55.51Samratanyone here tried freeswitch?
08:00.05drmessanowhy
08:00.30drmessanoThere is a very nice #freeswitch channel which offers lots of helpful support
08:00.40Samratjust wonder
08:08.41Samratdrmessano: thanks
08:09.59drmessanono probs
08:18.44mort_gibSamrat: People here prefer vanilla, you will find all kinds of compilations of "usefull" combinations.
08:19.30mort_gibIn my humble opinion they are really good, until you have to do x, where x is the function the companies behind didn't expect
08:20.09mort_gib-And if you "just run the CD" you end up not understanding what you are supporting
08:20.34mort_gibAnd that attitude should be left to the average MS Windows support technician
08:21.42Samratmort_gib: ok
08:22.05drmessanoWhat does that have to do with Freeswitch?
08:22.28cvnethas anyone here installed a2billing ?
08:22.53mort_gibdrmessano: Get off Coffee
08:23.11drmessanoHe said freeswitch, you idiot, not freepbx
08:23.17drmessanoGet off "stupid"
08:25.20mort_gibdrmessano: piss off
08:25.32drmessanomort_gib: learn to read
08:25.49mort_gib<PROTECTED>
08:31.17mvanbaakmort_gib: be nice
08:31.30mort_gibOk, sorry
08:31.56mort_gibSaturday, I'm in the office working and meeting clients in 20 minutes
08:32.22mvanbaakmort_gib: ugh, that sux
08:32.26mvanbaakit should be weekend
08:32.39mort_gibAnd hanging out in here I see the same shit, why do we need this crap: I wasn't being condescending.. that implies some level of passive aggression.  I was being outright insulting.
08:32.52mort_gibYes that sucks!
08:33.16drmessanoIt's called having a sense of humor, you stuck up, pompous jerk
08:33.45drmessanoStop taking IRC seriously and get some sun
08:33.56mort_gibStill, this was not a comment to you drmessano....
08:34.06drmessanoI dont care
08:34.36*** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net)
08:34.36mort_gibYou can quite happily piss off (sigh)
08:35.00drmessanoYou've told me that 3 times.... yet, not pissing off
08:35.03drmessanoMaybe you should give up
08:35.34drmessanoFact is, he asked about Freeswitch.. I kindly directed him to the proper location, you went on some rant about vanilla asterisk
08:35.40mvanbaakshows drmessano a picture of this 'lady' I saw on the street yesterday and watches him running off
08:35.43drmessanoStop being so rude to people
08:36.18mvanbaakshe was beyond ugly
08:36.18drmessanomvanbaak: Is she single?
08:36.33drmessanolol
08:36.37mvanbaaklike, 250kg, 1.90m and growing a beard
08:37.52drmessanoouch
08:38.19drmessanoDoes she have a 220v fondue pot?
08:40.37mvanbaakit was not in her shoppingcart.
08:40.52drmessanoHAW
08:40.59mvanbaakand I think everything she has was in that cart
08:41.44mvanbaakshe had clothes (or something she likes to call clothes) and some booze in there but that's pretty much it
08:42.49drmessanoProbably wild turkey
08:46.11drmessanoAnyone know the latest spandsp that works with 1.6?
08:46.23*** join/#asterisk axisys (n=axisys@bbgw10.bdcom.net)
08:46.48*** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk)
08:51.34*** join/#asterisk ibm2 (n=Administ@196.203.192.179)
08:53.37ibm2it's possible to activate xmpp in my switchvox
08:55.19mort_gibdrmessano: I mistook Freeswitch with one of the "compilations" your right, but did you ever have a look at it??
08:55.37drmessanoI know what Freeswitch is...
08:55.47mvanbaakdrmessano: 0.5
08:55.58mvanbaakdrmessano: I think there's a patch in trunk to support 0.6
08:55.59drmessanomvanbaak: TY, 0.6 is fail
08:56.02drmessanooh..
08:56.10mvanbaakso 1.6.2 will suppot 0.6
08:56.14drmessanoI see
08:56.20drmessanoIm ok with 0.5
08:56.33mort_gibDid you try it out??
08:57.18mort_gibAsterisk fork, yes but is it any good??
08:57.25drmessanoIts not an asterisk fork
08:58.11*** join/#asterisk piparkuka (n=igor@fw.wan.co.il)
08:59.17mort_gibThat's what this links says: http://www.freeswitch.org/node/117
09:00.23drmessanoThats not at all what that says, and its not a fork
09:01.23*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
09:01.33mort_gibI know, but did you try it out??
09:01.59drmessanoA long time ago
09:02.13mort_gib-Any good??
09:03.30drmessanoDunno.. I use Asterisk.. Looked interesting, I gave it a once over. Wasn't interested in learning another platform.  I also have a problem with how the product is positioned and the ANTI-asterisk sentiment that seems to come from the project.
09:04.48mort_gibI have head little about the project, I'm just curious, which is why I started looking at *
09:04.48drmessanoGenerally I don't see a lot of "Freeswitch is great, use it!".. I see "Asterisk sucks ass, ZOMG DIGIUM R TEH MICROSUK ZOMG, use this instead!!!11!!!"
09:05.11mort_gibOk, but it's NOT like * is perfect
09:05.12drmessanoSo I could care less about it at this point
09:05.20drmessanoNothing is perfect
09:05.30mort_gib:-) true
09:05.59drmessanoBut IMO, if Freeswitch was so great.. they could walk around exposing their giant hairy balls for all to see, not having to win people over who have problems with Asterisk
09:06.12drmessanoThat wreaks of "not good enough"
09:06.32mort_gibLike I have some 150+ * users on 8 different systems, and on ONE, only ONE has in frequent issues with dropped calls....
09:06.50mort_gibYeah, but the OpenSource camp is much like that
09:07.26mort_gibI use OpenBSD for some things, and it's really good, but damn those guys are slagging off EVERYBODY!
09:07.45drmessanoI dont see anywhere on Postgresql's site proclaiming "100 reasons you should use this instead of MySQL because MySQL sucks so hard, lemmetellya"
09:08.01drmessanoBSD users are zealots
09:08.03drmessanoThats different
09:08.05mort_gibFreeswitch actually has that??
09:08.14mort_gibI know, I still use OpenBSD
09:09.43drmessanoNo, but finding wholly qualified claims of Freeswitches superiority without being stacked against something in Asterisk are just about nonexistant.  I mean, I know "Asterisk" is the "ex-wife" to some of the devs.. But if you keep talking about how much of a bitch your ex-wife is, you're not gonnaa get a lot of dates
09:10.55mort_gibLOL, no that true!
09:13.25*** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net)
09:14.03kerxHi, is anyone around that knows about wholesale voip for broadcast dialing?
09:14.54mort_gibKerx: like Spamming
09:15.11kerxIs it considered spamming?
09:16.01mort_gibSounds a lot like spamming to me....
09:17.36kerxWell, I won't spend time trying to explain otherwise to you
09:17.47mort_gibNo don't :-)
09:17.49kerxI need VoiP minutes, do you know where I can go?
09:18.08mort_gibWell I use Voipon.co.uk
09:18.16mort_gibAnd others...
09:20.11kerxOk, I'll check it out thanks
09:20.38mort_gibThey accept IAX too, which is handy
09:20.39*** join/#asterisk Provito (n=Provito@pdpc/supporter/sustaining/Provito)
09:20.43drmessanomvanbaak: 0.5 FTW
09:20.44mort_gibSo you can use them in IAX Trunk mode
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09:30.16mvanbaakdrmessano: :)
09:30.23mvanbaakdrmessano: OpenBSD FTW
09:30.25mvanbaakhides
09:30.39drmessanozealot
09:31.05drmessanoI am going to write some code for 1.6
09:31.19drmessanoI wanna see if I can get app_fondue added to 1.6.3
09:31.43mvanbaakgheh
09:32.48ibm2can anyone tell me how i can activate xmpp standard  in my asterisk
09:35.20drmessanoWhat are you trying to do, ibm2?
09:35.58ibm2i try to activate IM between 2 bria
09:37.15drmessanoBria uses XMPP?
09:37.28ibm2yes
09:37.36drmessanoYou need an XMPP server then
09:38.35mvanbaakejabberd or something like that
09:38.35drmessanohttp://www.igniterealtime.org
09:38.35drmessanoGet openfire
09:38.43mvanbaakbut beware, the asterisk plugin is buggy at the moment
09:38.53drmessanoYes
09:38.55mvanbaakcant get the phonemappings page
09:39.14*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
09:39.14mvanbaakif I add an asterisk server it wont show up
09:39.14drmessanoWith 1.6?
09:39.16mvanbaakbut I do see it connecting to the asterisk AMI
09:39.17drmessanooh
09:39.26drmessanoI have a patch for that
09:39.33mvanbaakwhere is it ?
09:39.45drmessanohang on
09:41.28drmessanohttp://www.2l2o.com/asterisk/asterisk-im-WORKING.jar
09:41.45drmessanoThats the latest release with the db scripts fixed per a forum post
09:42.57ibm2yes i  already installed openfire
09:43.09mvanbaakdrmessano: cool, thanks
09:43.10drmessanook, and?
09:43.45ibm2but i have some problem with him
09:44.06drmessanoIf you cant get Bria talking to Openfire, you're wasting your time in here
09:45.44contactdqI need a dial plan that logs an agent in/out in 1.4. It would first prompt agent for agent no/pin. If agent was logged into that particular phone, it would log him out. If he was not logged into that phone, it would log agent out any extensions, and then log him into that phone. Anyone know where I could find something like this. I would be willing to pay a reasonable fee for this. It needs to be queuemetrics compatible and play the
09:45.55contactdqthanks
09:46.36drmessanoplay the thanks?
09:46.44drmessanoI R not understand
09:46.49drmessano:( meh
09:47.45contactdqlol
09:47.58contactdqno, agent logged in, agent logged off audio files :-)
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10:36.18contactdqanyone available to help figure out where I went wrong with a dialplan?
10:36.55mvanbaakjust pastebin the dialplan and a CLI log of when it fails
10:41.24contactdqhttp://pastebin.com/m140dc812
10:42.31contactdqPlaying 'cannot-complete-as-dialed'  in the CLI is all I get
10:43.11BBHosscontactdq: set debug 10
10:43.15BBHossset verbose 10
10:43.22contactdqok
10:43.32BBHossthen post cli, should be much more
10:44.20contactdqhttp://pastebin.com/m7a8f0436
10:44.50BBHosswhat are you dialing? 2001?
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10:45.28contactdqyes
10:45.34BBHossit looks like you're using freepbx, so i really can't tell you 100%
10:46.09BBHossif you were using asterisk, you'd need to make sure that the [macro-custom-agent-inout] is included in the from-internal context
10:46.48BBHossyou might try making an custom extension with the dial string of LOCAL/2001@macro-custom-agent-inout
10:47.01BBHossthen it would work from internal
10:47.01contactdqok
10:47.04var1hello, I'm about to download and compile Asterisk 1.6.0.1 if there is any reason I shouldn't speak now. ;)
10:47.19BBHossvar1: oh no wait, no!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
10:47.27BBHossvar1: j/k its fine :)
10:47.30var1lol
10:47.43var1thx BBHoss
10:47.48BBHosswelcome
10:48.36BBHossyou should get hossterisk 2.0 though, 100% erlang code with high avaliability and clustering built right into it from the ground up, all with gen_server/1
10:49.00BBHossdialplans stored in mnesia etc
10:49.25BBHossvar1: don't you want a link!?!?!
10:49.31var1yeah sure
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10:50.07BBHossheh, its just a dream right now, do you take dreammagnet links?
10:50.24var1I'm a bit overwhelmed with it all but I still take it in my stride :)
10:50.40BBHosshaha
10:50.55BBHosscontactdq: did that work?
10:52.37contactdqnope
10:52.44contactdqunfortuantely
10:53.57contactdqi'm trying to avoid agent_callbacklogin
10:54.01BBHosswhat did that log say with the custom extension, remember you must dial the custom extension number rather than 2001, unless you set the extension to 2001
10:54.31BBHosswell i really don't know jack shit about agents, much less in freepbx and its cronies
10:57.02contactdqwhat customn extension numbeR?
10:57.21contactdqsorry...i've always been able to bypass freepbx by using extensions_custom.conf etc
10:57.34BBHossthen why the hell use it in the first place
10:57.45BBHosssucks anyways, too bloated
10:58.32BBHosshow do you expect to be able to get to 2001 in the macro that you posted?
11:01.58contactdqi don't know.....
11:02.34BBHosscontactdq: you either need a custom extension, or you need to configure a number you can dial that will drop you in the macro that you posted
11:02.48contactdqok
11:03.34contactdqlet me create a custom extension
11:07.59*** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34)
11:08.38Dr-Linux|homeI want to discuss something about Queue application func... anybody ?
11:12.32Dr-Linux|homei want once a caller is waiting in the queue, moh should be played but once an agent/extension is chosen by the queue then caller should listen to ring. how i can do that?
11:13.28BBHossgoodnight all
11:14.35Dr-Linux|homegood noon to me
11:15.43BBHosswell technically its goo 6:15AM for me
11:20.58var1night  BBHoss!
11:39.20Dr-Linux|homevar1: around?
11:40.51var1around where, not here it's 12:40pm . so it's good afternoon really.
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11:42.16var1I don't normally hang out in irc but I will stick *around* here until I can't take anymore ;)
11:42.35var1that was a joke
11:43.43slingrlol
11:44.02slingri still haven gone to bed
11:44.09slingrits 7:40am for m
11:44.10slingre
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11:58.06kerxhi, can somebody please refer to me a good documentation to modify asterisk + astguiclient/vicidial from a predictive dialer to a survey based auto dialer?
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12:53.07Dr-Linux|homei want once a caller is waiting in the queue, moh should be played but once an agent/extension is chosen by the queue then caller should listen to ring. how i can do that?
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12:59.57johnakabeananyone know of anyway to spoof a provider to get unlimited channels
13:02.46johnakabeancome on, sip isn't that secure lol
13:06.51johnakabeanok, using Ulaw, how much bandwidth is needed up and down.......90 kbps??
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13:24.04donnibhi. what does 407 Proxy Authentication Required mean ?
13:24.15donnibi can't get inbound call to work
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13:36.40lmadsendonnib: it means the other end needs to authenticate, and that 407 has the nonce required to generate the MD5 hash in the 2nd INVITE from the device
13:36.49lmadsenINVITE (without auth) --->
13:36.56lmadsen<-- 407 Proxy Auth Required
13:37.03lmadsenINVITE (with authorization) -->
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13:37.44[gnubie]waves
13:39.41Dr-Linux|homei want once a caller is waiting in the queue, moh should be played but once an agent/extension is chosen by the queue then caller should listen to ring. how i can do that?
13:41.43[TK]D-FenderDr-Linux|home: this is documented in the sample configs... go read them...
13:42.12Dr-Linux|home[TK]D-Fender: tried but could't find
13:42.35[TK]D-FenderDr-Linux|home: Try again.
13:42.57Dr-Linux|home[TK]D-Fender: lemme open WIKI again
13:44.35[TK]D-FenderDr-Linux|home: "core show application queue" <-
13:44.46[TK]D-FenderDr-Linux|home: WIKI is the last place to look.
13:44.52Dr-Linux|homeok
13:46.24Dr-Linux|home[TK]D-Fender: it says:  'r' -- ring instead of playing MOH
13:47.30Dr-Linux|homebut that "rings" the queue from start unless call is assigned to an agent/ext but i want it should play music but once call is assigned to an agent/ext then I should start ringing
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14:58.39zchaoshey guys... i wanted to ask you what type of internet connection do i need to run a VOIP network? rogers has the following connections.... http://pastebin.com/m488ea235
15:02.18elielzchaos: it depends at least on the number of calls simultaneously that you want to pass throu the link and the codec that you will be using
15:03.27[TK]D-Fenderzchaos: Express minimum, keeping in mind Rogers sucks....
15:08.04*** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34)
15:08.15zchaosthis ins't a call centre
15:08.20zchaosits only going to be a call here or there
15:08.22zchaosthe odd call
15:08.26zchaosthere wont be many calls at all....
15:09.05zchaosthat should change the answer
15:11.29elielzchaos: there is a magnitude of 8 aprox. between using g729 (8kbps) and g711 (64kbps) so, it depends on your voip provider or the other end codec capabilities
15:13.19[gnubie]how do you usually interface or integrate existing legacy branded pabx to an asterisk box?
15:13.22*** join/#asterisk MalMen (n=Mal@bl8-126-62.dsl.telepac.pt)
15:13.29coppicethe bit rate ratio is more like 3:1 when you add in the overheads
15:14.05MalMenwhere can i find a list whith the best rated voip companys ?
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15:20.20edwin_quijadai am trying to compile zaptel and I get an error
15:20.30edwin_quijadaThe configure script was just executed, so 'make' needs to be
15:20.30edwin_quijada**** restarted.
15:20.45seanbrightdid you run ./configure first?
15:20.49edwin_quijadaand I get 1 error [config.status]
15:20.55edwin_quijadaseanbright: yes
15:21.11edwin_quijadaI get this when I do make menuselect or make
15:21.33seanbrightwhat version of zaptel?
15:21.35edwin_quijadahow can I unload a module ?
15:21.41edwin_quijadaseanbright: 1.4.12.1
15:22.00edwin_quijadai had a old version from zaptel running before
15:22.49seanbrightcan you pastebin your config.log?
15:22.52seanbright~pb
15:22.53jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
15:24.27edwin_quijadaseanbright:ok
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15:28.57edwin_quijadahttp://pastebin.com/m270d067a
15:31.27seanbrightwhat version of gcc do you have?
15:32.36edwin_quijadagcc version 4.1.2 20061115 (prerelease) (Debian 4.1.1-21)
15:32.44seanbrighthm.  strange.
15:33.34edwin_quijadawhat does mean?
15:33.44seanbrightnot sure.
15:33.50edwin_quijadait is the first time i get this
15:33.56RypPntry make clean ??
15:34.05RypPnthen ./configure
15:34.10edwin_quijadai did it
15:34.14edwin_quijadaget the same
15:34.35seanbrightmake distclean
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15:35.03tzafrir_laptopmake clean? that voodoo is not needed
15:35.12seanbrightannnnd here we go
15:35.14tzafrir_laptopjust: ./configure; make
15:35.23RypPn#
15:35.23RypPngcc: '-V' option must have argument
15:35.40RypPnmaybe the Makefile is wrong and it should be gcc -v
15:35.52seanbrightthat's in configure, not the Makefile
15:36.05RypPnwell, wherever it gets that from is wrong
15:36.17edwin_quijadatzafrir_laptop: i did it too
15:36.39tzafrir_laptopedwin_quijada, did the configure script end successfully?
15:36.50edwin_quijadatzafrir_laptop: yes
15:37.03tzafrir_laptopwhat's the output of:  make
15:37.13tzafrir_laptopcould you pastebin it?
15:37.16edwin_quijadaok
15:37.22edwin_quijadai reboot the pc
15:37.28edwin_quijadagive me a second
15:41.10edwin_quijadadid
15:41.17edwin_quijadai paste the last part
15:41.24edwin_quijadado u need everything?
15:42.16seanbrightpastebin all the output
15:43.04edwin_quijadahow can I unload all modules from asterisk?
15:43.32seanbrightone thing at a time
15:43.41seanbright'make' is not affected by what is running in asterisk
15:43.50seanbrightjust pastebin all the output when you run 'make'
15:44.59edwin_quijadaseanbright: I solved the problem.. dont kill me!!
15:45.07seanbrighti have no desire to kill you
15:45.17edwin_quijadajust the date was in the past
15:45.39edwin_quijadaso make launch error for the date ....
15:45.55edwin_quijadauhmmm... must say something about that??
15:46.07seanbrightyes... fix your system date
15:46.08edwin_quijadaWWell, thks! everyone!
15:46.11*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
15:46.26edwin_quijadai fixed the date now
15:46.32edwin_quijadaand compile fine!!
15:46.49seanbrightsuper.
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16:23.37zuezHey folks, is there any particular reason the following is not logging the userfield portion of my CDR? (CDR writing to database works wonderful otherwise, and the Verbose statement I put verifies ${PHONE_ID} is being populated: http://pastie.textmate.org/295315
16:24.12xacatecashi all, hope all is well.  astdb is persistent accross restarts, is there something similar which is not?
16:26.06xacatecasin particular I'm in need of some "storage" to store temporary presence information.
16:26.19[netman]global variables?
16:26.53xacatecasthat might work.
16:27.52xacatecasis it possible to construct those variables names dynamically?  in other words, if I have one channel variable that contains a string like "agent5", use that as an index into an array of sorts, or generate a variables name like "pres_agent5" from that?
16:28.13[netman]zuez: try using Set(CDR(userfield) in the h extension ...
16:29.29zuezso instead of exten => _., use exten => h, ?
16:29.52[netman]xacatecas: you can Set(pres_agent5,g) in any point of your dialplan
16:30.21[netman]yes zuez, that It was I said
16:32.36xacatecasnetman, agent5 is in a variable, so I can do Set(pres_${foo},g) ?
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16:33.17zuez[netman] that doesn't appear to do it either, oh well.
16:33.50seanbrightzuez: CDR backend are you using?
16:34.06zuezseanbright: cdr_addon.mysql.so
16:34.22zuezEverything is getting written to properly except userfield
16:34.26seanbrightwhat version?
16:34.52zuezseanbright: Asterisk 1.4.17~dfsg-2ubuntu1
16:35.13[netman]zuez: for that works, you first have to customize a cdr.conf setting, I remember
16:35.26seanbrightpastebin your cdr_mysql.conf file
16:35.32seanbrightmasking passwords where necessary
16:35.38zuezheh ok
16:35.44[netman]endbeforehexten=yes
16:37.40ManxPowerCould you have CDR batching enabled?
16:38.25seanbrightzuez: you need to add 'userfield=1' in the [global] section of cdr_mysql.conf
16:38.37seanbrightzuez: that will fix you up.
16:39.05zuezseanbright: thanks man. Let me try that.  is reloading just the cdr_addon_mysql.so module after modifying the config adequate for it to re-read the config into memory?
16:39.16seanbrightzuez: yes
16:40.39zuezseanbright: That did it, thanks.
16:40.44zuez[netman]: Thanks for the help as well!
16:40.50seanbrightzuez: no sweat.
16:40.50[netman]zuez: :)
16:40.55[netman]I forget that :(
16:41.46zuezI should update some of the wikis I've been finding by googling that don't make note of that, although it's probably expected of the user to read the sample configs and figure it out on their own, which I failed miserably at. :-)
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16:43.46Grnd-Wiregood morning everyone!
16:44.16seanbrightzuez: any expectation that the user will read anything is full hearty.
16:44.25seanbrighterr
16:44.28seanbrightfool*
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16:46.56ManxPowerzuez: see, if you had read the official docs none of this would have happened.
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16:47.27ManxPowerzuez: now is the time to go into your asterisk source dir, look in configs and doc
16:47.44seanbrightand the one in -addons as well
16:47.54ManxPowerthat too!
16:48.03Grnd-Wireooh, what did I miss?
16:48.19seanbrightnothing
16:48.40redder86After upgrading from zaptel-1.4.7 / asterisk-1.4.17 to zaptel-1.4.12.1 / asterisk-1.4.22 now all I get is a yellow alarm on my wcte1xxp.  Any ideas?
16:49.07Grnd-WireDid you update libpri to current as well?
16:49.07ManxPowerredder86: upgrading should not cause a yellow alarm.
16:49.31redder86Grnd-Wire: libpri 1.4.7
16:49.40Grnd-Wireok
16:49.42ManxPowerredder86: I assume you are using zttool to see the card status?
16:50.24redder86ManxPower: I was just saying what the CLI says from zap show status
16:50.45ManxPowerredder86: use zttool and have asterisk stopped.
16:51.03ManxPoweryour card should go green if you just have zaptel loaded.
16:52.00ManxPowerthat way you eliminate any issue with Asterisk or libpri and you can concentrate on zaptel
16:52.37redder86ok, I'm building zaptel with zttool now
16:54.25*** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk)
16:54.33redder86Alarms          Span                                               │
16:54.33redder86<PROTECTED>
16:54.37redder86coppice: hey
16:54.48redder86ManxPower: still yellow alarm
16:54.52ManxPowerredder86: have you tried a total reboot?
16:55.04redder86no, but I can do that now
16:55.41ManxPowerredder86: before you do that put a copy of your /etc/zaptel.conf on pastebin.ca
16:56.26ManxPowerredder86: are you physically near the box?
16:56.30redder86while it's rebooting... in full-disclosure it *is* an OpenVox D110PG.  zaptel is patched with their wcte1xxp.c changes, however
16:56.44coppiceredder86: hi
16:56.45redder86ManxPower: no, I'm 1000 miles away over an ocean.
16:56.49ManxPowerWell thank you for wasting my time.
16:57.02ManxPowerContact your hardware vendor.
16:57.37redder86ManxPower: ?
16:58.45seanbrighti thought it was pretty clear... :)
16:58.52ManxPowerredder86: I know nothing about that card, what patches it does, or how to diagnose problems with the card.  Go contact your hardware vendor for support.
16:59.24ManxPowerNow if you had a TE1xxP card, it would be a different thing.
16:59.41redder86sure, I was just hoping that someone here was willing to help troubleshoot regardless of the hardware
17:00.40Grnd-Wireredder86: I would try and get someone to physically unplug your T1 cable, let it go red.. wait.. and plug it back in..
17:00.53Grnd-WireYou can't do that over software.
17:01.07ManxPowerGrnd-Wire: I was going to suggest that if he was close to the system
17:01.30Grnd-WireManxPower: yeah - but you can find anyone who will follow instructions to do it.. Remote control monkey. :P
17:01.35redder86Grnd-Wire: I did that yesterday when someone was on-site
17:01.48Grnd-Wireredder86:ok, well then I am out of ideas
17:02.31ManxPowerredder86: in the future mention you have a non-Digium, non-Sangoma card right at the start.
17:02.53seanbrightthat way people can ignore you right from the get-go ;)
17:02.59seanbright(kidding)
17:03.00ManxPowerExactly!
17:03.05Grnd-WireManxPower: err.. Don't forget to mention Rhino
17:03.29ManxPowerseanbright: I'm not aware of even a single person using that card.  I doubt anyone here can help him other than the very generic stuff.
17:04.11seanbrightpoints to redder86
17:04.15seanbrightthere's a single person
17:04.19seanbright:)
17:04.20Grnd-Wirelaughs
17:04.23redder86I've used the card in a number of places.
17:04.39redder86I also have used many Sangoma cards
17:04.46redder86I've also used many Digium cards
17:04.58redder86I tend to be fairly open with what hardware I'll use.
17:05.16redder86I have had good support from all hardware vendors, including OpenVOX, it's just outside business hours right now.
17:05.23Grnd-WireOnce I find what works, I don't deviate..
17:05.39seanbrightclone hardware + custom patches = virtually impossible to diagnose what the problem is
17:06.23ManxPowerI'm here to answer *easy* questions, not here to answer difficult ones.   I answer difficult questions for money.
17:06.29drmessanoI was gonna buy a nxtvox card to play with.. the 4 port and 8 port base card are the same price... difference is the 8 has a patched Zaptel
17:06.35drmessanoNot worth it
17:06.40redder86seanbright: if you were to look at the patches you'd see that it's fairly straight-forward ... not much to be afraid of
17:06.48ManxPowerI pick a card vendor and stick with it.  Easier to keep spares around for one thing.
17:07.02seanbrightredder86: i'm not afraid
17:07.10drmessanoBesides, if you're using 8 analog lines, you're doing it wrong
17:07.20Grnd-Wireseanbright: oh yes you are - you're shaking in your boots
17:07.25seanbrightmaybe a little
17:07.31drmessanoseanbright: What is your name?
17:07.34drmessanoseanbright: What is your quest?
17:07.35redder86if I could standardize on Sangoma I would... however, some customers just can't stomach the price .... and I have to support hardware decisions made years and years ago that maybe I disagree with now
17:07.54ManxPowerredder86: then it sucks to be you.
17:08.03drmessanoseanbright: What is the zaptel timing ration of a disconnected TE100P?
17:08.05ManxPowerIn any case, BEST of luck with your card.
17:08.09Grnd-Wiredrmessano: I will never again use FXO on a new phone system. Too much risk of echo.. Gotta love the 4wire interface. :)
17:08.32drmessanolol
17:08.35ManxPowerGrnd-Wire: you realize echo comes from the FAR end analog loop, right?
17:09.04*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
17:09.07redder86ManxPower: I'm not afraid of spending money for support either... but your hostility is a total deal breaker.
17:09.29drmessanoManxPower: I think the point was you should have spent money on a better card
17:09.32drmessanoErrr
17:09.32ManxPowerredder86: I'd not take money from you anyway -- I'd not be able to do a good job of helping you.
17:09.36seanbrightthat's not hostility... it's charm
17:09.40drmessanoredder86: I think the point was you should have spent money on a better card
17:09.45Grnd-WireManxPower: err.. In the cases that I've had the problem, it appeared to be an impedance mismatch with my local loop. I had 6 lines, and each one performed differently.
17:10.00ManxPowerdrmessano: No, my point is that if you use a non-Digium/non-Sangoma card then you should not expect the community to be ABLE to help you.
17:10.17drmessanoSame difference lol
17:10.17ManxPowerGrnd-Wire: that is a possible cause of echo I guess.
17:10.22redder86sigh
17:11.05redder86nevermind.  I honestly was just trying to brainstorm here with people.
17:11.07Grnd-WireManxPower: yeah - You know, that whole hybrid matching, sidetone bullshit.. Which is why it's T1/PRI all the way for me now. :)
17:11.08ManxPowerhugs his Tellabs 96 channel T-2 echo canceller
17:11.24ManxPowerT-1 of course, not T-2
17:11.29Grnd-Wireheh..
17:11.50drmessanoredder86: IMO, being a tweaker and an experimenter, if you can MAKE something work, go for it.. if it's going 1000 miles away, spend money so the shit never crashes, pack it in a black box with no buttons, switches, or knobs, fill it with epoxy so it cant be fucked with, and send it off
17:12.21seanbrightthat is exactly how my russian bride was delivered.
17:12.26redder86drmessano: agreed 100% ... but the hardware was a decision made years ago under a different mindset
17:12.36*** join/#asterisk StephenF[W] (n=StephenF@198.144.197.28)
17:12.54drmessanoredder86: Sounds like deprecation time.. Load them up with a new card, take that card for something local
17:13.05xacatecas[netman], thanks, Set(GLOBAL(pres_${foo})=TRUE) will work.
17:13.19drmessanoPay a local tech to swap the card with a digium, config it remotely
17:13.19redder86drmessano: well, I'll just regress to the old versions
17:13.22seanbrightredder86: where is the box?
17:13.25ManxPowerdrmessano: or just get a customer that is not a cheap ass,.
17:13.47redder86seanbright: Hawaii
17:13.57seanbrightredder86: that's far from me.
17:13.59ManxPowerI do *NOT* recommend switching the card now.  That will just make things even MORE complicated.
17:14.01drmessanoManxPower: Even if they are, demand a new card.. demand something that somewhat sensible to work remotely
17:14.22redder86I won't be switching the card anyway.  I'll be just regressing versions.
17:14.26ManxPowerdrmessano: I enjoy doing tech stuff, not convincing the customer they are wrong.
17:15.03seanbrightredder86: is that patch known to work with 1.4.21.1?
17:15.05drmessanoManxPower: convincing?  Nah.. "If you want this fixed, this is what it will take.  Otherwise, it will remain broken."  Period.
17:15.36ManxPowerdrmessano: *nod*  Why not just not accept the customer in the first place.
17:15.37drmessanoCause and effect.. You paid me to fix it, I evaluated it, now here is my parts list
17:15.58drmessanoManxPower: If it was a problem from the onset, then set, I am 100% with you
17:16.03redder86seanbright: not sure there... the patch was made against 1.4.11 ... to which I'll be regressing, and then back to 1.4.7
17:16.07drmessanos/set/yes
17:16.11ManxPowerBTW, does anyone know of most recycling places accept old computers and monitors?
17:16.26seanbrightredder86: ahh.  lots has changed between 1.4.11 and 1.4.21.1... the patch applies cleanly though?
17:16.32ManxPower<PROTECTED>
17:16.38redder861.4.12.1 you mean?
17:16.43seanbrightyes, thank you
17:16.47redder86seanbright: yeah, the patch was clean
17:16.49drmessanoManxPower: No, but if you call your local county IT dept, they can likely point you to a place that does
17:16.58seanbrightbaltimore county does.  woo.
17:17.04seanbrightredder86: strange.
17:17.05drmessanoThey are most likely to know the responsible disposal path
17:17.10seanbrightredder86: might just be a bug in zaptel.
17:17.11ManxPowerdrmessano: I have an entire pickup load of old cases w/motherboards
17:17.36ManxPowerdrmessano: there's a dumpster on the property if all else fails they can go there.
17:17.46redder86seanbright: of course... which is why I kind of came here... but I'll backtrack versions now.
17:17.51drmessanoCall the county.. Should be easy enough
17:18.15seanbrightredder86: alrighty.  might want to check bugs.digium.com as well (if you care).  the bugs are tracked there.
17:18.20ManxPowerdrmessano: The country here doesn't even require proper disposal of used engine oil.  they say "soak it up with cat litter then throw it in the trash.
17:18.28drmessanolol
17:18.37redder86seanbright: bugs.digium.com is a joke as soon as you mention OpenVOX
17:18.52ManxPowerThe county has NO curb side pickup of recycleables
17:19.08seanbrightredder86: i wasn't suggesting opening a bug.  just taking a look to see if others are having yellow alarm problems with the latest and greatest.
17:20.01redder86seanbright: yeah, I may look into that if I have trouble regressing versions
17:20.27redder86seanbright: thanks for your open-mindedness
17:20.45seanbrightnods
17:22.02*** join/#asterisk xuser (n=JJ@unaffiliated/xuser)
17:22.24*** join/#asterisk StephenF[W] (n=none@198.144.197.28)
17:27.42loca|hosti have a big latency between the moment asterisk receive the call and the moment my sip phones (ring_all group) ring, some 5 seconds and that's a huge delay ... on the the flash panel, i can see the call is ringing on the trunk but transfered to the group after some time ... how to fix it ? same thing when the caller hangup, my sip phones continue ringing after that for some 5 seconds
17:28.22*** join/#asterisk VJFROMGT (n=vjfromgt@user-12lcpfg.cable.mindspring.com)
17:28.34VJFROMGT<PROTECTED>
17:34.07drmessano#mysql ?
17:34.10*** join/#asterisk StephenF[W] (n=none@198.144.197.28)
17:40.55loca|hostanyone ?
17:43.52*** join/#asterisk implicit (n=bayan@unaffiliated/implicit)
17:44.05jayteebeuller?
17:49.04ManxPowerloca|host: delays in incoming calls via analog FXO are usually caused by having asterisk configured to expect callerid info, but the telco is not sending the callerid info.
17:56.38*** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri)
18:01.01zchaoshey guys... i wanted to ask you what type of internet connection do i need to run a VOIP network? rogers has the following connections.... http://pastebin.com/m488ea235
18:01.28zchaosit isn't going to be a call centre , i just might get the odd call here and there... maybe 1 or 2 at a time at MOST ... but it wont happen often just the odd call
18:02.37drmessanoexpress
18:02.44[gnubie]how do you usually interface or integrate existing legacy branded pabx to an asterisk box?
18:02.50drmessanoLight MAY cut it, but not with any internet comms
18:03.08drmessano[gnubie]: T1 seems to be most common
18:03.18drmessano[gnubie]: T1 cards + T1 crossover
18:04.22[gnubie]drmessano: i see.. so the common legacy pabx out there comes with a t1 port already?
18:06.20drmessanoNope
18:06.20tzafrir_laptopwhat can I do if I have an IAX client that connects from behind a NAT and can't keep a steady port?
18:06.55drmessanotzafrir_laptop: Sounds like the router isn't handling NAT very well
18:07.30[gnubie]drmessano: so you mean, you have to add a t1 module on a legacy pabx first?
18:07.50drmessano[gnubie]: Lets do the math here.. A legacy PBX will have T1 or FXO, right?
18:08.08[gnubie]drmessano: yes
18:08.20drmessano[gnubie]: So those are your interface options.. Most serious PBX installing will be using PRI
18:08.36drmessano[gnubie]: IF you need to go FXO <> FXS you can, but man.. messy
18:08.42drmessanoAnalog = blah
18:08.57tzafrir_laptopdrmessano, some routers are known not to work very well. Which is why we have insecure=port in sip.conf
18:08.59drmessanoHow big is this OLD pbx install?
18:09.10[gnubie]drmessano: yes.. thanks..
18:09.44[gnubie]drmessano: i actually don't have an old pbx.. i just want to know how you integrate them
18:09.50[gnubie]thanks.. ;)
18:09.52tzafrir_laptopNAT routers in this case (two different cases) - sonicwall, fortinet
18:09.52drmessanotzafrir_laptop: If it's having that sort of problem with IAX, I dont see much you can do with it, other than replace the router
18:10.26drmessanohmmm
18:10.39drmessanoWhich sonicwall?
18:10.54drmessano[gnubie]: How big is the PBX install?
18:11.17tzafrir_laptopdrmessano, I'll have to check that (this is a case I recall from the past. and was not under my control, anyway)
18:12.19drmessanoIf its a TZ170, you can try new firmware.. SW is awesome about adding features, fixing bugs..
18:12.20*** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk)
18:12.46maxximhi. how can i create a varibale with a value in externsion file, in order to pass it to AGI script?
18:13.35maxximi was using:      exten => s,n,Set(DNID=91239123)
18:13.39maxximbut no effect...
18:17.22jayteemaxxim, try Set(CALLERID(dnid)=91238123)
18:18.21[gnubie]drmessano: i don't have a legacy pbx
18:18.33jayteeI do
18:18.46jayteeI even have a legacy car
18:19.07jayteeSubaru Legacy 1995 (last 5 speed manual front wheel drive they made)
18:19.14jayteeneeds a new water pump :-(
18:19.54drmessano[gnubie]: Then thats easy.. put a second NIC in your asterisk box, make a cat cable with a plug on one end, and at the other end, short 1 to 3, 2 to 6.
18:20.03drmessanoYou now have a crossover for your non=pbx
18:20.56[gnubie]i see..
18:21.10drmessanoOh
18:21.24drmessanoand if you want to tie in another legacy IP based PBX
18:21.42drmessanomake a SIP peer to 127.0.0.1 and a matching peer to 127.0.0.1
18:21.50drmessanoviola!
18:23.15jayteemy legacy pbx didn't do sip without an exhorbitantly expensive SIP ITG gateway card (Nortel) so I had to put Asterisk in between it and my telco on my T1 PRI spans.
18:23.24zchaosdrmessano so you think i need express for a little in home setup???
18:23.50drmessanozchaos: The offerings you showed me..
18:23.56zchaosyeah
18:23.59drmessanozchaos: the one below express was 1/256
18:24.09zchaosya
18:24.37zchaosthats to slow for a small basic voip setup?
18:24.37drmessanoI know for a fact you'll have problems with more than one call if theres ANY internet traffic.. I went through this not too long ago with a customer
18:24.49zchaosok
18:25.18*** join/#asterisk Marquel (n=Marquel@port-300.pppoe.wtnet.de)
18:25.22Marquelmorning
18:25.27zchaosis that because the d/l speeds or u/l speeds drmessano
18:25.32drmessanou/l speed
18:25.45zchaosexpress should be fine then eh
18:26.15drmessanoGetting above 256 is a huge hump
18:26.31zchaoshmm
18:27.07drmessanoYoure the one with the home based busines, right?
18:27.49Marquelis there a possibility to play a sound to both legs of a call, after the called party picked up, before bridging them?
18:28.03drmessanoSo you've got your business calls and occasional home call, correct?
18:28.32drmessanoLets do some math here
18:28.45drmessanoWhat are the prices of the 1MB and 7MB, the two we are stuck on
18:29.06[gnubie]waves to all.. gtg now.. thanks..
18:29.25drmessanoand answer my first question.. Are you gonna use the VOIP for the home business and POTS for the home line, or going all SIP, or what?
18:29.34drmessanoI dont remember all the details
18:31.44drmessanozchaos?
18:36.41*** join/#asterisk seanmh (n=seanmh@c-68-35-21-64.hsd1.nm.comcast.net)
18:39.55*** join/#asterisk sucituanbo (n=free@c-24-21-121-148.hsd1.wa.comcast.net)
18:42.44[TK]D-Fender~iwmwb
18:42.44jbotI WANT MY WEEKEND BACK!
18:45.03drmessanoI was going to give him a sensible plan
18:45.37drmessanoLike buying G729 licenses, and saving some bacon if he's not gonna use that connection for much
18:45.46drmessano1MB/256k would be enough
18:46.41zchaossorry drmessano
18:46.46zchaoshad to dos omething int he kitchen
18:47.09zchaossorry drmessano please continue to help me haha
18:47.12drmessanoSo what is the plan for the comms
18:47.14zchaosyeah im' the guy with the home business
18:47.17drmessanoSIP, POTS?
18:47.35zchaosso i will get the odd calls for the business and home
18:47.48zchaosprobably only 2 calls at once, once in a blue mooon
18:47.49drmessanoBoth over SIP?
18:47.54zchaossorry what is sip again
18:48.04drmessanoAre you going IP or getting phone lines?
18:48.17zchaosphone lines
18:48.22zchaosi'm going ot order 2 phone lines
18:48.26zchaosand use ata boxes
18:48.28ManxPowerzchaos: You should read the Asterisk Book before you do anything else.  You will learn a lot.
18:48.33drmessanoSo bandwidth is irrelevant
18:48.48zchaosso i dont need express?
18:48.49drmessanoHang on
18:48.56zchaosdrmessano
18:49.00zchaosi have a diagram one sec
18:49.04drmessanoTwo COPPER LINES?  or two DIDs from a provider?
18:49.06ManxPowerwhat is an express?
18:49.17drmessanoService level from Rogers cabel
18:49.23drmessanocable
18:49.29ManxPowerAh.  The poor sod.
18:49.35drmessanoHes got 4 options..
18:49.52drmessanoThe two I am teetering on are 1MB/256k and 7MB/512K
18:50.02drmessano256k is weak
18:50.05ManxPowerI'll bet they don't do CPC
18:50.06zchaoshttp://imagebin.org/28979
18:50.17zchaosthe only difff thing is
18:50.20zchaostehre are going to be 2 ata boxes
18:50.23zchaosone for each phone line...
18:50.30zchaosthats the only thing missing form the diagram
18:50.44ManxPowerzchaos: those are NOT lines, they are DIDs
18:50.51drmessanoWait
18:50.55drmessanoThe ATAs go to the PBX
18:51.07ManxPowerin any case until you read The Book you're screwed and it is pointless to try to help.
18:51.55drmessanoIf you're getting 2 DIDs from a VoIP provider, you are using SIP
18:51.59drmessanoand that will go to the PBX
18:52.01zchaosmanxpower
18:52.03zchaosi have someone setting it up
18:52.09zchaosi was just trying ot confirm what connection i need
18:52.10zchaosthats all
18:52.15drmessanoIs that the diagram they gave you?
18:52.24zchaosnah i did it with some assistance
18:52.32drmessanoBecause the flow is all wrong
18:52.38[TK]D-Fendercompletely
18:52.44*** join/#asterisk beek (n=klinebl@65.211.106.242)
18:52.46zchaos..... lol
18:52.48zchaoswhats great
18:52.49zchaossigh
18:53.09zchaoschange the ata boxes to the pbx?
18:53.11[TK]D-Fenderzchaos: If you have someone setting all this up, let them do their job.
18:53.22zchaosi'm doing the hardware
18:53.25zchaoshe will do the software
18:55.05ManxPowerThis channel is too Monday'ish right now.  See y'all later.
18:55.08*** part/#asterisk ManxPower (n=manxpowe@208.sub-75-200-201.myvzw.com)
18:55.38[TK]D-Fenderzchaos: Actually your picture is fine.
18:55.42drmessanozchaos: What service level do you have now?
18:55.46zchaoshttp://imagebin.org/28980
18:55.49zchaosi changed the diagram
18:55.51zchaosfender you sure?
18:55.56zchaosthe guy helping me was sure my diagram was right too
18:55.59zchaoshe said it wuold work fine
18:56.04drmessanoGet a 2 port ATA
18:56.07drmessanoDont get 2 ATAs
18:56.12drmessanoThats a huge waste
18:56.19drmessano1 Port cost more than the 2
18:56.28drmessanoSPA-2102
18:56.28zchaosi was told i need 2
18:56.30drmessanoNo
18:56.32drmessanoYou dont
18:56.32zchaosso the pbx konws how to handle
18:56.34zchaosthe calls
18:56.37drmessanoNo
18:56.41drmessanoYou were told wrong
18:57.16zchaosuhmmmm
18:57.26drmessanoThey make 8 port ATA's and 24 port channel banks (big ATA) that don't confuse anything
18:57.35drmessanoSo thats incorrect
18:57.46[TK]D-Fenderzchaos: a single 2-port ATA = $50.  And screw the part of the diagram showing that calls will forward to cell, etc from that side of things.  Call flow is handled by the PBX, it is not an extension of your phone end-points on the ATA
18:58.48zchaosso was hte picture correct hte first time?
18:59.36drmessanono
18:59.47drmessanoATA was going to the router
18:59.51drmessanoThat is NOT CORRECT
18:59.58[TK]D-Fenderdrmessano: It can be for sure
18:59.59drmessanoATA to the PBX
19:00.02*** join/#asterisk moy (n=moy@189.169.85.251)
19:00.16zchaosargh
19:00.17drmessanoNot for call flow
19:00.18[TK]D-Fenderdrmessano: it is pgoing to be physically wired to the SWITCH on th router
19:00.34drmessanook
19:00.50zchaossooooo
19:00.52zchaosis it fine? lol
19:01.10[TK]D-Fenderzchaos: http://imagebin.org/28979 <- this is fine for WIRING, and drop that "cell" bit off the right side
19:02.08[TK]D-Fenderzchaos: And don't call one port "land-line business", and the other "home phone".  these sound like 2 different animals.  you plug a PHONE onto an ATA port.  The fact you want 1 to ring for calls concerning BUSINESS is irrelevent
19:02.30zchaoshttp://imagebin.org/28981
19:02.32zchaosthere.....
19:02.50zchaosfender i know
19:02.52zchaosi just did it for me
19:02.53zchaosthats all
19:03.02[TK]D-Fenderzchaos: What is this magical term "land line" doing on a phone connected to an ATA?
19:03.34zchaosa stationary phone thats all i mean by it
19:03.45[TK]D-Fenderzchaos: they are BOTH stationary phones
19:04.01zchaosits just wording everyone konws what i mean
19:04.21[TK]D-Fenderzchaos: When nothing you say adds up, no, we don't
19:04.33zchaoshttp://imagebin.org/28982
19:04.35zchaosthere....
19:04.46[TK]D-Fenderzchaos: Who can trust someones understanding when they terms are passed through a blender?
19:05.03*** join/#asterisk kamanashisroy (n=kamanash@119.30.35.36)
19:05.34zchaosthere
19:05.57[TK]D-Fenderzchaos: Better.  No you can have * ring whatever phones you want for any given call.  You can also choose what outbound resources to use for each port on that ATA.
19:06.00[TK]D-FenderNow*
19:06.47loca|hosti have a big latency between the moment asterisk receive the call and the moment my sip phones (ring_all group) ring, some 5 seconds and that's a huge delay ... on the the flash panel, i can see the call is ringing on the trunk but transfered to the group after some time ... how to fix it ? same thing when the caller hangup, my sip phones continue ringing after that for some 5 seconds
19:07.07[TK]D-Fenderzchaos: ATA's do not have any association to DID's, lines, or any other resources.  All they are are independant SIP devices who's calls are processed by the PBX in whatever way you tell it to.
19:07.18cvnetanyone here used asterisk2bill ?
19:07.31zchaosok
19:08.46zchaosok so its all good now
19:08.56zchaosi think the other guy was tyring to tell me
19:09.02zchaosthe reason he wanted 2 atas
19:09.14*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
19:09.25zchaoswas because * wouldn't knbow how ot distinguish which line is ringing and how to handle it without 2 atas
19:10.39maxxim$AGI->exec('UserEvent','_SIP_Auth|User-Name:\ '.$UserName."|".'DNID:\ '.$dnid."|".'Channel:\ '.$input{'channel'});
19:11.02maxximunfortunately, UserEvent is not splitting the variables in Manager. why?
19:11.49zchaosfender 2 more questions
19:11.57zchaos1) is this the one i should ordre - http://shop.gloria.ca/voip/adapters/voip-2-port-fxs-analog-adapter/prod_5352.html
19:12.23zchaos2) back to my original question... so what type of connection speed do i need for this? http://pastebin.com/m488ea235
19:12.32maxximit goes like a strigth line: UserEv UserEvent: _SIP_Auth|User-Name: uknorth|DNID: 91239123|Channel: OOH323/h323gw-eec7
19:12.59[TK]D-Fenderzchaos: Each port is 100% separate from the other and call control is always handled by *.  If you want 500 phones to ring after * accepts the inbound call, so be it
19:13.17zchaosgotcha
19:13.22[TK]D-FenderczhAnd if your other guy doesn't know this then you might want to find someone else who does
19:14.06[TK]D-Fenderzchaos: Yes that ATA is fine
19:14.44[TK]D-Fenderzchaos: Express.
19:15.27zchaoswould be nice if i could find a damn provider in canada who carries the SPA2102
19:15.28zchaossheesh
19:17.51zchaosknow of any good voip hardware providers?
19:17.57zchaosgoogle is coming up dry
19:18.02zchaosfor canada of course
19:18.22[TK]D-Fenderhttp://www.canadianvoipstore.com/home.php
19:18.42[TK]D-Fenderhttp://www.voipdepot.ca/
19:18.55[TK]D-Fenderhttp://www.voipware.ca/
19:19.27maxximwhat is the right version of UserEvent application for * 6 ?
19:19.28zchaoscanadavoip store wants $25 ins hipping lol
19:19.30zchaosi saw all those places
19:19.45maxximwhat is the right version of UserEvent application for * 1.6.0 ?
19:20.02maxximwhat is the right usage (can you give me an example) of UserEvent application for * 1.6.0 ?
19:20.30zchaosthanks fora ll yoru help fender
19:20.54[TK]D-Fendermaxxim: Is your AGI wrapper eve 1.6 compatible?
19:21.31zchaosfender i'm confused with these 2 products....
19:21.32zchaos1) http://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=90
19:21.40zchaos2) http://www.voipdepot.ca/index.php?main_page=product_info&cPath=1&products_id=79
19:21.46zchaosi dont need one wtih a router
19:21.52zchaosis #1 the same as #2 without hte router?
19:22.04*** join/#asterisk maxxim (n=maxxxim@93-96-96-106.zone4.bethere.co.uk)
19:22.10maxximsorry, got a disconnect
19:22.24maxximi've tried directoly in extension file. got the example from wiki
19:22.29maxximUserEvent(ASTDB|Family: dnd|State: on)
19:22.42*** join/#asterisk tkbeat (n=tk@p54B962F4.dip.t-dialin.net)
19:22.55maxximthe problem is that in Manager it shows variables in one row
19:23.19maxximbut in the definiton of userEvent, it says that | should split variables
19:23.24[TK]D-Fenderzchaos: 2102 includes a router, T.38 support, bigger CPU, etc.  Worth a few extra $
19:23.47zchaospoint noted
19:23.48zchaosthanks
19:23.51[TK]D-Fendermaxxim: Again, are you even sure your AGI wrapper is 1.6 compatible?
19:24.45maxxim[TK]D-Fender> exten => s,n,UserEvent(ASTDB|Family: dnd|State: on)
19:24.55maxxim[TK]D-Fender> i'm using directly, withou AGi
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19:26.27[TK]D-Fendermaxxim: Ok, looks fine. pastebin some usable backup.
19:27.39maxxim[TK]D-Fender> sorry, what to put in pastebin?
19:27.59[TK]D-Fendermaxxim: AMI dumps of message attempts, CLI output from when you issue it, etc
19:29.57maxxim[TK]D-Fender> http://rafb.net/p/TgNfH846.html
19:34.26maxxim[TK]D-Fender> any other output? just tell me how
19:34.39[TK]D-Fendermaxxim: Not sure... reading up on it now.
19:37.58maxximmanager_event(EVENT_FLAG_USER, "UserEvent", "UserEvent: %s\r\n%s", args.eventname, buf);
19:38.20maxximwho should split | , manager_event function, or in app_userevent.c ?
19:39.08[TK]D-Fendermaxxim: Not sure, and I can't see any obvious error...
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19:42.35maxxim[TK]D-Fender> are you still, looking? what shoul I do?
19:43.43[TK]D-Fendermaxxim: no idea
19:44.28maxxim[TK]D-Fender> have you ever used this UserEvent 'application'?
19:44.51[TK]D-Fendermaxxim: not personally
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20:03.54*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
20:09.42lesouvageI have to check an Asterisk server that has trouble at random. Now and then the person that has been called doesn't hear the caller. With the next try/call it is working ok. Server is in operation for a long time without problems. Any suggestions about what can cause this kind of problems. I doubt it has to do with the Asterisk server itself.
20:09.50*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
20:16.03*** join/#asterisk fatherx (i=RCP@abraham.sh.cvut.cz)
20:17.00lesouvageI noticed I didn't ask a question. Does any of you has  suggestions about what can cause this kind of problems?
20:27.31lesouvage.
20:29.05lesouvagehas the world finally come to an end to make space for a interstallair bypass ?
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20:35.46sdanielswhat is the network topology like? Iv seen this sort of thing wiith bad DSPs in a media gateway.
20:35.58jmb287hi, i'm looking for some help with Ast 1.6.1 and problems with MWI
20:37.02Dr-Linux|homeI'd like to ask same question again:
20:37.03Dr-Linux|homeactually i want when a caller is waiting in the queue, he/she listen to a MOH but as that call is assigned to an agent/ext then caller should listen to a ring
20:38.13jmb287i've got a linksys spa 942  and Asterisk 1.6..  Message Waiting seems broke, any thoughts on how to troubleshoot
20:43.10lesouvagesdaniels: I just have access to the asterisk server. Personally I suspect the SIP provider.
20:44.01lesouvagejmb287: what do you mean with "Message Waiting seems broke"?
20:44.59jmb287lesouvage:  I leave a vm and the phone light doesn't come on.   I power cycle the phone and the light comes on.   I delete msg and the light does not go off, power cycle phone and light goes off
20:45.36sdanielslesouvage: well your problem is that one of the RTP streams is jacked up for some reason, there isnt really anything I can tell you without knowing the full path for each call leg..
20:45.59lesouvagejmb287: sorry, I'm not a blinking light expert.
20:46.36sdanielslesouvage: I suggest that if the asterisk box is on a managed witch that you set up a port monitor and wireshark it to see whats going on. id also wireshard the port the phone is on.
20:46.50sdanielsthis keyboard sucks
20:47.17jmb287lesouvage:  you might also want to make sure that there isn't a firewall jacking with the RTP stream
20:49.12lesouvagejmb287: I suspect them to have change some network components or settings of the network components like the firewall without having their voip solution and sip trunk into consideration.
20:51.06jmb287yup, time to wireshark the link, or run tcpdump on the * box
20:52.42SQLDarklyHas anyone noticed when using realtime sip they do not show in the CLI when using "sip show peers/users" but still register? Is this an asterisk bug or am I misssing something
20:53.54lesouvagesdaniels and jmb287: thanks for the input, you have been really helpfull
20:54.34jmb287No worries.  back to bashing my head on this stupid MWI issue, grrrr
20:58.45maxximUserEvent application in 1.6 doesn't split body parameters like in 1.4 * .Could anyone help me find out why?
21:04.35hesco.
21:04.42lesouvagesdaniels: isn't a media gateway with DSPs a typical SIP telco device?
21:05.27*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
21:05.40hescoI'm getting lots of cdr results looking like this:  duration: 88, disposition: NO ANSWER, lastapp: Hangup
21:05.45sdanielslesouvage: Its a Cisco router that has voice DSPs in it to convert TDM to G711 or whatever codec you want.
21:06.26hescohow does that work? Hangup follows the playing of a 48 second message.
21:07.04sdanielslesouvage: usually you would get something like a 3825 and put a bunch of PRI wics in it, this would be your gateway to the PSTN from your netowrk
21:07.15lesouvagesdaniels: The kind of device that is used by SIP providers to internconnect to the pstn network?
21:08.15*** join/#asterisk LiNeTuX (n=LiNeTuX@253.238.95.24.cfl.res.rr.com)
21:09.30*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
21:12.12*** join/#asterisk sbc383 (n=stuff@S0106000fea3b527f.cg.shawcable.net)
21:12.33sdanielslesouvage: well yeah probably that too.. but you could call ATT or whoever and tell em you want a couple of t1s to the PSTN then plug em into the 3825, then plug your core router into the 3825 via ehternet and hang your asterisk box and phones off of the core router.
21:12.58sdanielspstn---3825----yournetwork
21:13.41sbc383I'm quite new to asterisk, and have a question that I can't find the answer to anywhere. In the situation where an outside caller reaches the Asterisk PBX and hits an extension to dial an internal phone, is it possible to first play a message to the inside caller, then bridge the channels?
21:13.48lesouvagesdaniels: yes I know, it is actually what is the "normal" scenario in Europe with 1 or more E1 .
21:14.35lesouvagesbc383: exten => s,n,Playback(beep)   ; or any other message
21:15.26hescohow does that work? Hangup follows the playing of a 48 second message.  Where does that disposition string come from and why does it interpret an answered call that runs to completion as having not been answered?
21:15.37Dr-Linux|homeactually i want when a caller is waiting in the queue, he/she listen to a MOH but as that call is assigned to an agent/ext then caller should listen to a ring
21:15.45Dr-Linux|homeis that possible?
21:16.01lesouvagesbc383: exten => s,n,Answer() before the Playback
21:16.05maxximUserEvent application in 1.6 doesn't split body parameters like in 1.4 * .Could anyone help me find out why? Where can i report this bug?
21:16.06[TK]D-Fendersbc383: the A() option for Dial.
21:17.00sbc383[TK]D-Fender: hehe, right in the man pages, doh! thanks dude
21:18.12*** join/#asterisk ManxPower (n=manxpowe@208.sub-75-200-201.myvzw.com)
21:21.52[TK]D-Fendersbc383: when in doubt : "core show application dial"
21:23.47jmb287anyone around that can help with a MWI (message waiting indication) prbm ?
21:24.58maxxim[TK]D-Fender> where shoul i report this issue with userevent?
21:25.13[TK]D-Fendermaxxim: Mantis
21:25.47maxxim[TK]D-Fender> can you give me pls more details, where is URL for mantis?
21:28.26ManxPowerbugs.digium.com
21:28.37jmb287bugs.digium.com  is not working  :(
21:29.08jmb287i get a tcp connection reset when trying to reach the bugs site
21:29.50ManxPowerjmb287: then you'll have to wait.
21:42.54Dr-Linux|homeManxPower: moving around for a long time but my questoin is still there :P
21:43.04*** join/#asterisk BBHoss (n=bbhoss@c-68-62-170-33.hsd1.al.comcast.net)
21:43.31Dr-Linux|homeactually i want when a caller is waiting in the queue, he/she listen to a MOH but as that call is assigned to an agent/ext then caller should listen to a ring
21:46.40*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:54.09*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
21:54.25lesouvage.
21:56.20jeevsup foo
21:56.25jeevgot any $ left over from SNL ?
21:58.51jmb287looking for help with a Message Waiting issue  MWI  ???
22:09.14*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
22:14.39*** join/#asterisk errr (n=errr@fedora/errr)
22:21.48Qwelldrmessano: wake up
22:23.31Dr-Linux|homeactually i want when a caller is waiting in the queue, he/she listen to a MOH but as that call is assigned to an agent/ext then caller should listen to a ring
22:24.38mvanbaakQwell: let him enjoy weekend
22:24.41QwellNO!
22:24.45Qwell:D
22:24.51mvanbaakmeh
22:25.02Qwellmvanbaak: how's it going?
22:25.07Dr-Linux|homeso any advice for that, if that is possible or not?
22:25.24mvanbaakgood :) almost ready for a week of painting, redoing bathroom etc
22:25.37Qwellwoot
22:25.49Qwellalready moved?
22:25.53mvanbaaknext weekend we'll move into our new house
22:25.56Qwellahh
22:26.10mvanbaaknah, painting ceiling and walls is easier when there's no stuff in the new house
22:26.14jmb287anyone here that can help with a MWI issue ?
22:26.50mvanbaakone more week and we're moved \o/
22:27.24mvanbaakQwell: Nov 29th party at my place
22:27.35QwellI'll be there!
22:27.38Qwellwell, I'll be here
22:27.41Qwellbut I'll be there in spirit
22:27.47mvanbaak:)
22:28.24mvanbaakI'll dcc you a beer from time to time
22:29.00mvanbaakmy new house rox, specially the kitchen
22:29.32mvanbaakQwell: please reboot bugs.digium.com
22:29.45Qwellmvanbaak: I think only russellb can
22:29.48mvanbaakhere's a pic of my new kitchen:
22:29.51mvanbaakhttp://picasaweb.google.com/mvanbaak/NewHouseDenHaag#5257486581735917090
22:30.51*** join/#asterisk Defraz (n=T0tal@63.228.246.250)
22:32.52mvanbaakQwell: then call him to do it !
22:32.53mvanbaak;)
22:33.07Qwelljust sent him an sms actually
22:34.06jayteethat is an awesome kitchen
22:34.48*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
22:34.48*** mode/#asterisk [+o russellb] by ChanServ
22:35.07*** join/#asterisk moy (n=moy@189.169.70.164)
22:41.18mvanbaakhey russellb !
22:41.25russellbhi
22:41.33mvanbaakbugs. is down
22:41.36russellbi know
22:41.39russellbthat's why i'm on IRC
22:41.41russellbtrying to work on it
22:41.51mvanbaakpoor you. even in the weekend we need you
22:41.58russellb:)
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22:44.26*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
22:48.16drmessanobugs is down?
22:48.17drmessanoHmm
22:48.30drmessanoDid someone submit a bug report about that?
22:49.13russellbQwell did
22:49.17russellbhe submitted the bug report via sms
22:49.18russellb:-p
22:49.22Qwellto russellb
22:49.26Qwell<3
22:54.02*** join/#asterisk kerx (n=prepro@adsl-69-104-18-9.dsl.irvnca.pacbell.net)
22:54.07kerxhowzde
22:56.33kerxhi, can somebody please help me?
22:56.45kerxhaving an issue w/ a sip call
22:57.06kerxdo you know if i am using NAT, I have to do any port-forwarding setup's on the router for SIP to establish properly?
22:58.07mvanbaak_~sipnat
22:58.07jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
22:58.34kerxThanks
22:58.40mvanbaakTHIS IS NOT THE TIME TO GO TO SLEEP !
22:58.41mvanbaak;)
22:58.52kerxWhat ?
22:59.06kerxConfused me :)
22:59.06mvanbaakkerx: nothing. ignore that ;)
22:59.09kerxok
22:59.10kerx:)
22:59.26mvanbaakkerx: it's because of the quit message of russell
22:59.37kerxdo we have pastebin here?
22:59.43kerxoh i see ! :!!!!
22:59.46kerxslaps himself
22:59.47mvanbaak~pastebin
22:59.47jbot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
22:59.55kerxoo, nice list
23:00.15kerxhttp://pastebin.ca/1230531
23:00.17kerxcan u please check that out
23:00.20kerxsomething weird is going on
23:01.32kerxOh wow
23:01.36kerxOct 18 19:09:39 DEBUG[4968] chan_sip.c: Registration successful
23:01.41kerxfirst time I saw that in the log file :)
23:01.53kerx1 sip peers [1 online , 0 offline]
23:01.54kerxnice
23:03.48mvanbaakhhmm, that pastebin is not enough info to find the problem
23:03.59*** join/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
23:03.59*** mode/#asterisk [+o russellb] by ChanServ
23:04.08kerxwb russellb who goes to sleep
23:04.10mvanbaakwb poor russellb
23:04.20kerxmvanbaak, what else do u need, i can try to give u everything :)
23:04.30mvanbaak<--- feels sad that russellb has to work in the weekend
23:04.50mvanbaakkerx: try to make the call with 'sip set debug on'
23:05.07kerxNo such command 'sip set' (type 'help' for help)
23:05.45mvanbaakkerx: ok, try: sip debug
23:05.56kerxok, i did
23:06.07kerxvici*CLI> sip debug peer gafachi
23:06.07kerxSIP Debugging Enabled for IP: 64.192.112.13:5060
23:07.00kerxok, got it
23:07.35kerxhttp://pastebin.ca/1230532
23:08.54mvanbaakcan you pastebin your sip.conf ?
23:10.08kerxyep
23:10.13kerxhttp://pastebin.ca/1230533
23:10.47kerxouchie, there he goes again :P
23:11.04mvanbaakand your box is behind nat ?
23:11.30kerxyep, my ip is 192.168.1.2 (asterisk machine), and 192.168.1.1 is the linksys nat
23:12.05mvanbaakkerx: then you need the 'externip' and 'localnet' settings in sip.conf
23:12.11kerxoh
23:13.09kerxin general or gefachi?
23:13.32mvanbaakgeneral
23:14.00kerxok, i added those, and i did a  'sip reload'
23:14.38kerxlet me try now
23:14.52kerxsame
23:14.53kerxOct 18 19:23:02 NOTICE[9343]: pbx_spool.c:269 attempt_thread: Call failed to go through, reason 8
23:15.25[TK]D-Fenderkerx: Read up :
23:15.27[TK]D-Fender~sipnat
23:15.27jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
23:15.30[TK]D-Fender^^^^^^
23:15.40mvanbaakkerx: try this: http://michiel.vanbaak.info/temp/sip_nat_settings_6.txt
23:15.42kerxu sure it's a nat issue at this point?
23:16.04[TK]D-Fenderkerx: you aren't set up properly for it at all
23:16.11kerxwhy?
23:17.25kerxmvanbaak, i used the script, and plugged in those settings
23:17.27kerxexternip = 69.104.18.9
23:17.27kerxlocalnet = 192.168.1.255/255.255.255.0
23:17.29kerxsip reload
23:17.32kerxand it did same thing
23:17.33[TK]D-Fenderkerx: Contact: <sip:a7281st6JLPt1tEj@192.168.1.2> <-- because you did not tell * what local subnets you have and what your WAN IP is.  This contact header shows that you haven't done it right.  Now go read the guide
23:18.35kerxhttp://pastebin.ca/1230538
23:18.38kerxcan you please check this one
23:18.42kerxi've done the correct settings now
23:18.44kerxand it still fails
23:19.56mvanbaakkerx: SIP/2.0 404 Not Found
23:20.38kerxhrmm.... why would that happen?
23:21.40mvanbaakshow us the dialplan
23:22.44*** join/#asterisk freakazoid0223 (n=mattc@68.162.74.19)
23:22.45sdanielsQuick question, with asterisk are the RTP streams always routed though the asterisk server?   pone--asterisk--phone or does asterisk tell the phones to connect directly?
23:22.47kerxhttp://pastebin.ca/1230542    < - extensions.conf
23:23.00kerxhttp://pastebin.ca/1230543    < - callme.call   (my test call file that goes into /var/spool/asterisk/outgoing)
23:23.07mvanbaaksdaniels: depends on the setup
23:23.21sdanielsmvanbaak: whats default?
23:25.56mvanbaaksdaniels: see sip.conf section ';----------------------------------- MEDIA HANDLING --------------------------------'
23:26.55kerxso any idea on my weird setup :)
23:27.10kerxit's most likely i have a feeling that I don't understand callback files and how they work w/ the dialplan correctly
23:28.01sdanielsOK Ill take a look, the reason I ask is because I'm interested in having a phone register to an * box that is behind a firewall, if you forward 5060 to the * box, then it seems that all RDP will have to go through the * for this to work.
23:28.08*** join/#asterisk mog (n=mog@c-68-62-219-86.hsd1.al.comcast.net)
23:28.08*** mode/#asterisk [+o mog] by ChanServ
23:28.31kerxsdaniels, I haven't done any Port Forwarding to he * box though
23:28.35*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
23:28.52kerxwb
23:29.18mvanbaak_sdaniels: you saw my lines ?
23:29.42sdanielsmvanbaak: ???
23:30.09mvanbaak_01:23 <       sdaniels> mvanbaak: whats default?
23:30.09mvanbaak_01:25 <       mvanbaak> sdaniels: see sip.conf section ';----------------------------------- MEDIA HANDLING --------------------------------'
23:30.12mvanbaak_01:26 <       mvanbaak>  ; By default, Asterisk tries to re-invite the audio to an optimal path. If there's
23:30.15mvanbaak_01:26 <       mvanbaak>  ; no reason for Asterisk to stay in the media path, the media will be redirected.
23:30.50mvanbaakmy connection died
23:30.59kerxhehe
23:31.12kerxstupid ghost connection's
23:31.15kerxi hate 'em
23:31.25mvanbaakyup
23:31.37mvanbaakmy connection is shaky the last couple of days
23:31.40sdanielsmvanbaak: oh yes, i saw that, thx
23:31.52mvanbaakand I'm not going to fix it. because I'm moving to a new place next week
23:32.39kerxnice
23:32.48kerxmvanbaak, did you have a chance to check out my dialplan, did it look ok?
23:32.59kerxhttp://pastebin.ca/1230542    < - extensions.conf
23:33.01kerxhttp://pastebin.ca/1230543    < - callme.call   (my test call file that goes into /var/spool/asterisk/outgoing)
23:35.54mvanbaakkerx: it's pretty simple
23:36.03kerxyeah it looks like it's that 404
23:36.09mvanbaakoutboundmessage1 has no exten => 18183459801
23:36.11kerxnow i saw a 401 unauthorized in my SIP establishment
23:36.46kerxbut it still shows the peer is online and established w/ me
23:36.55kerxwhy would i need a exten =>1818..
23:37.03kerxthat's my phone number, isn't that supposed to come from the callback file?
23:37.19mvanbaak#
23:37.20mvanbaakContext: outboundmessage1
23:37.20mvanbaak#
23:37.20mvanbaakExtension: 18183459801
23:37.20mvanbaak#
23:37.22mvanbaakPriority: 1
23:37.43kerxok?
23:37.58mvanbaakthat's where you want to go to with the callfile
23:38.07kerxoh
23:38.20kerxi should replace those s's with the number huh?
23:39.25mvanbaaktry it and see
23:39.31kerx<-- SIP read from 64.192.112.13:5060:
23:39.31kerxSIP/2.0 404 Not Found
23:39.33kerxdamn :-(
23:39.43kerxName/username              Host            Dyn Nat ACL Port     Status
23:39.43kerxgafachi/a7281st6JLPt1tEj   64.192.112.13        N      5060     Unmonitored
23:39.43kerx1 sip peers [1 online , 0 offline]
23:39.46kerxmy peer's are online!
23:39.50kerxwhy it can't find it?
23:40.11mvanbaakit does find the peer
23:40.31mvanbaakBUT
23:40.42mvanbaakyou call the peer, without telling the peer what number to reach
23:40.46mvanbaakI think that's the problem
23:41.10mvanbaakgafachi is a provider ?
23:41.11kerxoh
23:41.13kerxyes sir
23:41.17kerxthat could be the problem
23:41.27kerxi don't know too much w/ call files and the dialplan
23:41.41kerxthe whole time i thought extension was the phone number :-(
23:41.47mvanbaakso you try to setup a call to the provider without telling the provider who/what to call
23:42.06kerxwhere do u specifyc where to call?
23:42.13mvanbaakin the callfile
23:42.34mvanbaakSIP/gafachi/<your_cellphone>
23:42.37kerxyea,is that info supposed to be in the Channel: ?
23:42.40mvanbaakor something like that
23:42.43kerxoh
23:42.48kerxlet me try that
23:43.28kerxshit
23:43.29kerxthat was it :)
23:43.33kerxi hear the phone ring
23:43.39kerxhugs mvanbaak!
23:43.47mvanbaakthere you go
23:43.52kerxit didnt play the sound though!
23:44.04kerxthat's another thing :)
23:44.10mvanbaakyup
23:45.28kerxyeah, that text2wave is not even executing properly
23:48.53sdanielsAny opinions on the best sip trunk provider?
23:50.08kerxso far i've been hearing information that gafachi (one i've signed up with) is pretty good
23:50.23kerxbut ive heard  broadvox  is the best
23:50.34kerxor i should say, one of the best
23:50.43mvanbaak~
23:50.48mvanbaak~siptrunk
23:50.49jbotNo such thing, my friend.. Like too much salty plum soda.
23:51.05mvanbaak~trunk
23:51.06jbotmethinks 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.
23:52.40kerxwhen u modify the dialplan in extensions.conf do u have to reload asterisk?
23:52.56sdanielsno you can reload the dialplan
23:53.01kerxok
23:53.16sdanielsjust dialplan reload
23:53.43mvanbaakor for older versions of asterisk: extensions reload
23:54.42kerxthx
23:54.47kerxi'm running the older it seems
23:55.53kerxyeah, definitely something is wrong with the sound coming out of my * machine
23:56.17mvanbaakwhat version is it ?
23:56.46kerx1.2.27
23:57.40kerxso besides the nat settings in sip.conf i don't have to setup port forwarding for an * client connecting to a sip server?
23:57.49kerxnow i'm stuck at the audio not coming out of the machine (*)
23:58.25mvanbaakfirst of all, upgrade to 1.6
23:58.46[TK]D-FenderNo, first of all, fix your NAT settings
23:58.58mvanbaakmeh
23:59.17mvanbaakman, I really should start upgrading my boxen
23:59.17kerxTK: I've done all the NAT settings finally :)
23:59.25kerxI installed w/  Vicidialnow
23:59.29kerxThat's why I haven't upgraded yet
23:59.42cvneti just installed vanilla fresh, edited my sip.conf and added user 100 and 101 (didnt touch the extention.conf) should i be able to connect to the box from outside network?
23:59.51[TK]D-Fenderkerx: Go follow the guide.
23:59.56kerxWhich guide?

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