IRC log for #asterisk on 20090926

00:05.18manxpower~answers
00:05.19infobotextra, extra, read all about it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
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00:55.01jelly-bean1i have a poly 320 and 330 and 501. the 320 and 330 work file downloading their firmware and getting IPs etc. the 501 cannot even get an IP via DHCP. any ideas what i have to do to the 501? i have *468 already
00:55.46thehar~switchvox
00:56.13manxpowerjelly-bean1: usually that means you pluged the lan cable into the PC port on the phone
00:56.36jelly-bean1hehe i did do that once to the 320 but i just double-checked and its in the LAN port
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00:57.00raden_work[TK]D-Fender, dude reg issues with vitelity again tonight :(
00:57.24raden_workrestarted asterisk didnt go away
00:57.51raden_workrebooted router same BS
00:58.03raden_workshutdown asterisk for 5 min all was good
00:58.12manxpowerI never have registration issues with Vitelity
00:58.24manxpowermaybe your router just sucks?
00:58.32raden_workis there a way when a registration fails for it to take like a 2 min break before it tries again
00:58.41raden_workmanxpower, ill know when i get my asa 5505 in place
00:58.52raden_workI think part of this is router related
00:59.23manxpowerraden_work: I assume you have qualify=yes for that peer.
00:59.27raden_workthere any downfall to stacking HP procurve switches
00:59.36raden_workmanxpower, yes but the registration itself is failing
00:59.43raden_workwas failing
00:59.57manxpowerraden_work: go to wal-mart.  spend $60 on a linksys
01:00.16manxpower1) the problem is fixed or 2) you now have a spare "router"
01:00.28manxpowersounds like win-win to me
01:02.43manxpowerjelly-bean1: there are only a few things that could be wrong.  The connection from the phone to the DHCP server might have a problem (wiring) or the boot setup is set to use a specific VLAN or is set to use CDP and you don't have CDP setup.  After checking those things it's time to RMA the phone and get on with your life.
01:02.49raden_workmanxpower, I have a Cisco ASA-5505 here just need to get it configured
01:03.10manxpowerraden_work: you have spent weeks on this problem.
01:03.29jelly-bean1manxpower: it is set to use CDP. wtf is cdp?
01:03.30raden_workmanxpower, yes i know :(
01:03.48raden_workmanxpower, goal next week get cisco in place
01:03.49manxpowerjelly-bean1: CDP = method to automatically select the right VLAN for the phone to use.
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01:03.51raden_workthen groupwise
01:04.00raden_workthen ups software for linux
01:04.01raden_workthen done
01:04.30raden_workgrr i have to go bbl
01:17.33jayteeloves CDP
01:17.50manxpowerjaytee: me too, but if you don't understand it......
01:18.02thehargiggles
01:18.21jayteealso loves Option 66 for DHCP to point to my server that handles FTP provisioning
01:19.06jayteenow if I can just get SIP TCP working with Exchange Unified Messaging I can dump running sipX on a VM
01:19.10manxpowerjaytee: me too.  Deploying a phone should just require plugging it into the LAN and let the network take care of it.
01:20.06jayteemanxpower, yeah well I've written a bash script that copies config templates and uses SED to edit them with parameters I pass on the command line
01:21.04jayteeso for a 330 I just type ./prepphone.sh "MAC ADDRESS" XXXX   and I'm done where XXXX is the 4 digit phone extension
01:21.06manxpowerjaytee: I'm working on a cgi script to parse an uploaded .CSV and bulk build extensions and config files
01:21.26jayteeI'm working on a web interface to run the scripts from PHP
01:21.47manxpowerjaytee: no matter what I do I always seem to get dragged into FreePBX
01:22.22jayteenot me, I'd rather write my own custom GUI. I've got too many things in my dialplan that FreePBX would kill
01:22.54manxpowerjaytee: Unfortunately I can't dictate what a customer gets anymore since I'm not a consultant anymore
01:24.06jayteethis is due to setting up my dialplan originally to be able to have 4 digit dialing between my now decommissioned Nortel Option 11C and Asterisk and modeling certain Nortel features in the * dialplan.
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02:16.43DJCharlieEvening all. I have a small question about setting up a dialplan. I think I have it laid out properly, but when I try choosing from my main menu, I get hung up on with an error of "WARNING[4974]: pbx.c:2437 __ast_pbx_run: Channel 'Zap/1-1' sent into invalid extension '2' in context 'incoming', but no invalid handler"
02:17.02DJCharlieA log (and part of my dialplan) are at http://pastebin.com/d2920025c
02:17.10DJCharlieCan someone help me fix this?
02:19.51DJCharlieHello?
02:21.46*** join/#asterisk SirThomas_Home (n=tomc@209-169-199-174.mn.warpdriveonline.com)
02:24.27DJCharlieAnyone awake?
02:26.21*** join/#asterisk scalex000 (n=chatzill@50puntacana02.codetel.net.do)
02:26.33scalex000good evening!
02:26.42DJCharlieEvening. :)
02:26.44scalex000I have couple question about sip.conf
02:26.56scalex000who can help me?
02:27.01DJCharlieJoin the club. I'm still waiting.
02:27.12scalex000let me paste the packet
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02:35.41scalex000http://pastebin.ca/1579866
02:36.04scalex000line 53
02:36.59DJCharlieWish I could help you scale, but I'm still trying to figure out a dialplan.
02:37.15DJCharlieAnd I don't believe anyone else in here is really here.
02:37.50scalex000they are here, maybe very tired to type.
02:38.16superbeefit's crazy its like people might do othe rshit on a friday night
02:38.40DJCharlieWell, I don't understand why anyone would idle in a support channel of all places.
02:39.19superbeefsupport channel or not, IRC is historically a presence based environment
02:39.33superbeefbesides people can scroll back and look for something interesting if they have to
02:40.33DJCharlieWell would you care to scroll back and see if you could help scale or myself please?
02:43.43superbeefso it's choking on this?
02:43.44superbeeflets call Pennsylvania Pennsylvakia
02:43.46superbeefno
02:43.48superbeefwrong paste lol
02:43.55superbeefit's choking on this
02:43.56superbeefexten => t,2,Goto(menu2,2,1)
02:43.57superbeefright?
02:44.00scalex000superbeef: do you know where I can put how to interconnect BCM and asterisk
02:44.13superbeefwhat's BCM?
02:44.28scalex000superbeef: nortel BCM
02:44.40superbeefscalex000: probably a SIP trunk
02:44.42DJCharliesuperbeef: Actually, it chokes no matter which menu option I choose.
02:45.24scalex000superbeef: I want to share my experience ohh days and days reading so another not take too long to do it
02:46.35superbeefDJCharlie: so what's this guy doing exactly
02:47.09superbeefscalex000: not working so easily eh?
02:47.20DJCharliesuperbeef: From the first answering message [incoming] you should be able to dial 2, 4, 5, 9, or 777.
02:47.41DJCharlieBUT, no matter what you dial, it hangs up immediately, with that error message.
02:47.48scalex000superbeef: well for now yes
02:48.28scalex000superbeef: I have a little inconvinience when dial a mobile phone through BCM
02:49.31scalex000superbeef: If I need to dial 11 digit, nortel not say anything only ring and ring
02:49.35superbeefDJCharlie: which line is supposed to listen for input for digits?
02:50.04DJCharliesuperbeef: Doesn't Background() do that?
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02:54.29superbeeftry changing exten => t,2,Goto(menu2,2,1) to exten => t,2,Goto(menu2,s,1)
02:54.33superbeefhttp://www.voip-info.org/wiki/view/Asterisk+tips+ivr+menu
02:55.28superbeefscalex000: what's the call path like?
02:55.49DJCharliesuperbeef: No joy, same error.
02:56.36scalex000ok
02:57.18superbeeftry changing exten => t,2,Goto(menu2,2,1) to exten => t,2,Goto(menu2,s,2)
02:58.19DJCharliesuperbeef: But that will point it to Read(REPEAT) and not to the outgoing message for menu 2.
02:58.26superbeefscalex000: it's okay?    lol... like what's the topology[expected] of a complete call
02:59.38scalex000superbeef: what do you need exactly? I don't know what to paste or say.
03:00.08superbeefex: sip phone -> Nortel -> SIP trunk -> asterisk -> T1 PSTN
03:00.40superbeefDJCharlie: are you sure?
03:01.02scalex000superbeef: well I create a extension _8X9XXXXXXX, 1, Dial(SIP/81${EXTEN},45)
03:01.12superbeefDJCharlie: either your goto is amiss, or the say you're defining the extensiion in the macro is amiss
03:01.37scalex000superbeef: well I create a extension _8X9XXXXXXX, 1, Dial(SIP/81${EXTEN}@outbound-asterisk,45)
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03:02.00superbeefscalex000: lets simplify and think about the flow of the call outside of asterisk syntax
03:03.09scalex000superbeef: :(
03:03.49superbeefscalex000: lol common man, you can't articulate how you want it to work?
03:04.03DJCharliesuperbeef: I'm certain. Just tried it.
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03:04.18scalex000superbeef: No I not american people
03:06.53scalex000superbeef: its working now
03:07.14superbeefscalex000: cool.. how did you fix it?
03:08.13Sandheaverfound an IRC log of this channel for the past 8 years.  reading it is like speed-lurking.
03:08.17scalex000superbeef: I mean i can dialout through nortel without problem, but when i need know if nortel is saying a message asterisk not hear
03:09.24superbeefscalex000: you want nortel to call asterisk?
03:09.58superbeefSandheaver: pipe it through grep and wc and see how who says the most profanities
03:10.14scalex000superbeef: well I connect Asterisk to Nortel and is already don
03:10.16scalex000done
03:11.00superbeefscalex000: see if you can say this a different way "if nortel is saying a message asterisk not hear"
03:11.33scalex000superbeef: right, you are in the point
03:12.03scalex000for example: the number you have dial is disconnect at this moment
03:12.17scalex000superbeef: asterisk only ring and ring
03:13.13superbeefscalex000: what does /var/log/asterisk/full say when its ring ring?
03:13.24superbeefDJCharlie: don't stop believing
03:13.30scalex000superbeef: I don't know
03:13.38scalex000superbeef: ha  ha ha
03:14.04superbeefscalex000: Are you allowed to see the logs?
03:14.52scalex000superbeef: which one?
03:14.59superbeefscalex000: asterisk
03:15.43superbeefscalex000: you need to see the asterisk logs when you Nortel -> Asterisk
03:16.47scalex000superbeef: I mean what filename you need to see, I have 4
03:18.01superbeefscalex000: you need to watch /var/log/asterisk/full while you call asterisk with nortel.... (tail -f /var/log/asterisk/full)
03:18.29superbeefscalex000: we want to know what asterisk sees when nortel calls
03:19.06scalex000superbeef: well I not have full file I have messages
03:19.47superbeefscalex000: pastebin.com
03:20.26DJCharliesuperbeef: Any more ideas?
03:22.05scalex000superbeef: too long
03:22.30superbeefscalex000: you need to find the part this is just frmo when you call
03:22.33superbeefscalex000: not the whole file
03:22.53superbeefscalex000: you need to "see" in the logs where nortel is ringing asterisk
03:23.05superbeefscalex000: those errors will help you figure out what is happening
03:23.16scalex000ok
03:26.05superbeefDJCharlie: running out of steam sorry
03:27.26superbeefslumber
03:27.57DJCharlieAnyone else?
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03:28.52scalex000superbeef: http://pastebin.ca/1579898
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03:31.48superbeefscalex000: is nortel a phone or a PBX?
03:32.17scalex000superbeef: PBX
03:34.14superbeefscalex000: that's tough
03:34.17superbeefhttps://issues.asterisk.org/view.php?id=15102
03:34.27superbeefscalex000: i have to sleep...good luck
03:34.45scalex000superbeef: thank you for your time
03:34.53scalex000superbeef: see you later
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04:17.38radenanyone have logs that can tell me who was helping me with my asa 5505 the other night i dont know if i was on as raden or raden_work
04:20.18radenhello
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04:26.10radenSandheaver: ?
04:26.21Sandheaveryes?
04:26.28Sandheaveroh asa
04:26.38Sandheaverwhat's up
04:26.44Sandheavertry virtualization?
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04:39.05Sandheaversleepytime
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04:46.07psilikonexit
05:08.00radenlol
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05:29.35verywisemancan i build load-balance cluster for asterisk?
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05:38.28drmessanoum yeah
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05:42.39Grofneed help
05:42.45Grofasterisk is dropping SIP call
05:47.40TJNIIGrof: Post some details or you'll never get an answer.
05:53.56*** join/#asterisk Zackery (n=me@74-131-188-77.dhcp.insightbb.com)
05:55.46p3nguinWhat exactly is SIP forwarding?  Does it require a trunk from one ITSP to another or what?
05:57.14nextime'morning
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06:37.30Grof[Sep 26 08:27:52] DEBUG[30200] chan_sip.c: Got unsupported a:fmtp:101 0-16 in SDP offer
06:37.30Grof[Sep 26 08:27:52] NOTICE[30200] chan_sip.c: No compatible codecs, not accepting this offer!
06:37.32Grof?
06:38.04Grofwhat is a:fmtp:101 ?
06:38.10Grofand why does it bother asterisk?
06:45.00kaldemarGrof: http://www.rfc-editor.org/rfc/rfc2198.txt
06:48.49coppiceGrof: 0-16 probably means 101 is RFC2833. if that's the case * shouldn't complain
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06:51.21Groftnx
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07:30.05sunzofmancalls are being rejected b/c extensions are not found.
07:32.39sunzofmanextensions.conf -> http://pastebin.com/d1ee8951a
07:34.36kaldemarsunzofman: what is calling into asterisk? what context have you defined for the caller? what number is called?
07:36.03kaldemarthese questions would be answered by a CLI output of a call and sip.conf.
07:36.49sunzofman<PROTECTED>
07:38.40kaldemarCLI output of the whole call, not a single line. it was clear that you don't have extension 3132630611 in your dialplan. and why are you calling yourself?
07:40.10[TK]D-Fendersunzofman: And you clearly have no match forr that #
07:41.24sunzofmankaldemar: just wanted to test my new install
07:41.26kaldemardeja vu, btw
07:42.04sunzofmansip.conf -> http://pastebin.com/d50f627c3
07:42.45sunzofmankaldemar: the output shared from CLI was the only output which appeared after the call was made.
07:42.59kaldemarsunzofman: kind of a useless test, use app Echo for example.
07:43.34kaldemarsunzofman: "core set verbose 10" and try again. you'll get more output.
07:44.23sunzofmankaldemar: ok
07:49.00kaldemarhmm. i remember telling you to add a from-broadvoice context in your dialplan earlier this week.
07:50.36sunzofmankaldemar: must i replace the existing context=internal under [general] in my dialplan?
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07:51.27sunzofmankaldemar: ok i see add the [from-broadvoice] context and then populate the necessary action under neath
07:51.36kaldemarsunzofman: no, replace the context under general with something that leads to a dead end. use valid context for defined peers only.
07:52.10kaldemarbut yes, add [from-broadvoice] to extensions.conf
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07:59.17[TK]D-Fender[03:40]<[TK]D-Fender>sunzofman: And you clearly have no match forr that #
07:59.43regan40hi
07:59.48kaldemarand i clearly don't have a pastebin to look at.
08:00.33sunzofman[TK]D-Fender: I obsfucated the number in the earlier pastebin.
08:00.53sunzofmankaldemar:sip.conf -> http://pastebin.com/d50f627c3
08:01.09sunzofmankaldemar: extensions.conf -> http://pastebin.com/d1ee8951a
08:02.09kaldemarsunzofman: you posted those already. i'm waiting for the CLI output of a call.
08:02.19[TK]D-Fendersunzofman: context=from-broadvoice <--- and where is this context in your DIALPLAN?
08:03.44sunzofman[TK]D-Fender: i just added it to extensions.conf
08:04.27[TK]D-Fendersunzofman: I don't see that
08:08.11regan40I know it is probably a FAQ but what cards are recommended for asterisk,,,>
08:08.14regan40..?
08:09.32[TK]D-Fenderregan40: Digium & Sangoma are the front runners
08:09.32regan40looking for analog cards and australai PSTN compatible
08:09.54regan40googles Digium & Sangoma
08:10.28sunzofmankaldemar: latest CLI output -> http://pastebin.com/d85edb9e
08:11.15sunzofman[TK]D-Fender: i didn't repost extentions.conf after the new context [from-broadvoice] was added. i'll do that now.
08:11.41kaldemarsunzofman: that's not the whole call
08:12.10[TK]D-Fender'3132630611' <- nothing to match this
08:12.49sunzofmankaldemar: verbosity is set to 10, that is the resulting output.
08:13.53kaldemarsunzofman: go add extensions to broadvoice-in
08:16.15[TK]D-Fenderkaldemar: I'll leave this one to you.. best of luck with that...
08:16.29sunzofmankaldemar: so, [from-broadvoice] context should have 'exten=> 101,x,x' and exten=> 102,x,x' underneath..
08:17.32kaldemarsunzofman: if you're going to dial 101 and 102 from something that uses the context, yes.
08:18.02kaldemarsunzofman: but those still won't match 3132630611.
08:19.09sunzofmankaldemar: gotcha.. regarding matching the bvoice #.. not sure how to do that. the number behaves as an authentication component.
08:19.33sunzofmankaldemar: bvoice requires this authID.
08:20.14kaldemarexten => 3132630611,1,...
08:22.56sunzofmankaldemar: 3132630611 is the # that allows the outside to reach my internal extentions..
08:24.10kaldemarsunzofman: the call comes in by that number, and you need to send it somewhere. you do that with e.g. exten => 3132630611,1,Dial(SIP/101)
08:25.01sunzofmankaldemar: understood
08:29.04sunzofmankaldemar: updated extension.conf -> http://pastebin.com/d305907d0
08:33.25sunzofmankaldemar: still stubbing my knuckles ;-)
08:34.24sunzofmankaldemar: i've not changed sip.conf
08:36.31kaldemardid you call in already?
08:39.57sunzofmankaldemar: yep.. i get a different CLI output and the generic bvoicemail is gone. i'm making progress the [from-broadvoice] context seems to have fixed it.
08:40.12sunzofmankaldemar:  -- Executing [3132630611@from-broadvoice:1] Answer("SIP/3132630611-7c416000", "") in new stack
08:40.13sunzofman<PROTECTED>
08:43.36*** join/#asterisk trentster (n=marks@d58-110-225-234.sun6.vic.optusnet.com.au)
08:45.14trentsterHey have setup asterisk on a intel nano platform, if at compile time I choose all the audio greetings in all the formats like g729 etc will asterisk automatically choose the corerct version and not transcode greetings etc/
08:45.18trentster?
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09:12.19*** join/#asterisk TheOpenSourcerer (n=TheOpenS@81-178-65-1.dsl.pipex.com)
09:12.33Miladhi, is any command to transfer a call ?  I have extension number and channel  name and wana transfer this call to another extension
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09:44.16GuggeMilad transfer maybe?
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09:45.33MiladGugge, I check out transfer, but I should call it in dialplan, but I want put a button on 3party app and when click that transfer call, so I think I need another command to run az AMI
09:47.52*** join/#asterisk znh (n=hans@unaffiliated/znh)
09:47.58znhHello.
09:48.32znhI configured two phones, one hard the other soft. If I call the echo service I won't hear echo from the hard phone
09:49.18znhAny ideas?
09:49.38znhI can ring the hardphone from the softphone, but also no sound
09:49.56znhthe hardphone is connected via a router. on the same lan
09:50.02znhboth SIP
09:50.12GuggeMilad, i press transfer on my phone, and then the number to the phone i want to transfer to
09:50.13Guggethats it
09:53.58MiladGugge, yes I know that way , I want to create another way to transfer call for our CTI
09:55.48Miladanyway tnx for your clues
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10:49.50dustybinsomething strange is going on, if i ring my polycom using my softphone, my polycom makes the ringing sound, if i dial in from outside, my polycom is silent, only a green light flashes to indicate the call
10:50.25dustybini think i know what it is
10:56.08dustybinyep, i knew it, i have a virus
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11:28.52DonAlexHey guys....
11:29.55DonAlexcan someone help me debug why the gui is playing up. I have tried running asterisk -d -vvvvv etc. but I cannot see any debug information at all.. yet I know it is listening to port 8088 because lsof -i tcp:8088 shows it is..
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11:30.47DonAlexThe symptoms are that no matter what url I put in is cannot seem to find any files.
11:30.54MiladDonAlex, can you run asterisk -r ?
11:31.00DonAlexyes
11:31.08Miladand you have a console ?
11:31.12DonAlexI am running it in debug mode atm
11:31.39Miladow -d option :) yes
11:31.52DonAlexyes I have a console.
11:32.03DonAlexcore ser verbose 5 shows nothing either..
11:32.14DonAlexI justr cannot see any http request at all..
11:32.33DonAlexeven though I know it is accepting them because of the error pages..
11:33.07MiladDonAlex, do you test core set debug 10 ?
11:33.23DonAlexcan do.. will that show http stuff?
11:33.54Miladow sorry you want to use GUI
11:34.43DonAlexyes.. the server is running but I cannot figure out what is going on.. the files all seem inthe right place.. right permissions but no url can access any files.. so I want to debug what is actually going on
11:35.03DonAlexam I really going to have to use strace? ;)
11:35.08DonAlexbloody hope not :P
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11:35.27Miladye ! always my problem solve with strace ;)
11:35.54DonAlexYeah but on Asterisk ? multithreaded as it is? *whimpers*
11:39.09DonAlexIs there really no way to debug the http server from within asterisk?
11:39.16DonAlexsurely that cannot be right?
11:40.55DonAlexhmmmmm
11:41.01DonAlexsomething does not look quite right here..
11:41.24DonAlexhttp://pastebin.com/m6916c3cc
11:41.54superbeefwhat doesn tlook right in that
11:42.05DonAlexnow I know there is no phoneprov configures.. so that is why there is .. before the path.. but why does static have that too?
11:42.26DonAlexalso the default dir is static-http
11:42.33DonAlexcould it just be misnamed?
11:44.07DonAlexhmm the manager url works..
11:44.54DonAlexas does the mxml one..
11:44.56DonAlexhmmm
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11:48.44DonAlexOk so where the hell are those default url defined?!
11:48.49DonAlexin the source?!
11:55.15zambai have a cisco ata-188.. when dialing internal extensions (only four digits), it takes several seconds before the numbers are sent to the * and actually dialed
11:55.30zambadoes anyone know where i can tweak this in the cisco adapter?
11:55.31DonAlexDOH!!!!
11:55.32DonAlexok ok..
11:55.34DonAlexsolved it..
11:55.45DonAlexGood old Debian shifting things around again.. ;)
11:56.13DonAlexby default asterisk-gui  puts static-http into /var/lib/asterisk..
11:56.47DonAlexhowever the ast_config_AST_DATA_DIR is /usr/share/asterisk in Debian.
11:57.00DonAlexmade  a symlink and the files are found.. Cool.
11:57.01DonAlex;)
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12:18.49znhI configured two phones, one hard the other soft. If I call the echo service I won't hear echo from the hard phone
12:19.00znhAny suggestions?
12:21.53zambai have a problem with my adapter.. i can only call devices attached to it for the first few minutes after registering.. then i'm unable to dial them.. my adapter is a ata-188
12:22.24zambathe adapter is behind NAT
12:22.36zambait works just fine with my softphones
12:27.03Guggeset the adapter to register every minute or something like that
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12:40.51zambaGugge: yeah, i did.. that worked
12:40.58zambaGugge: but what causes it?
12:41.23zambaGugge: and why don't i get the same behavior with my softphones?
12:49.37*** join/#asterisk Adr3nalin3 (n=afink@204.26.87.226)
12:49.50Guggeyou nat device closes the forwarded udp port after xxx seconds idle
12:50.01Guggemaybe your softphone sends something every xxx seconds
12:50.06Guggeand your adapter dont
12:50.16Guggea register renews the udp forward
12:50.22Gugge(any traffic does)
12:50.34Guggeso another sollutions is to make a new call every minute :)
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12:59.28sircolinI likes jux
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13:12.02zambaGugge: hehe
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13:24.48znhHello.
13:24.57znhI configured two phones, one hard the other soft. If I call the echo service I won't hear echo from the hard phone.
13:25.41znhsip show peers shows the phones are connected. The hardphone has my external IP (phone is connected to internet router). Others have an internal ip address.
13:25.52znhHas that anything to do with it?
13:26.11znhGugge: any idea?
13:27.15znhIt's not a codec issue for sure. I forced a codec which is supported
13:32.24znh"exited non-zero on" is the error message if I hang up
13:40.58Guggefix it so you hard phone has an internal ip too
13:41.24Guggei guess you asterisk cant send the sound back to the phone when its shown as the external ip
13:41.35Guggebut without total knowledge about you network setup, its only a guess
13:42.05znhYes I figured that asterisks connect to the external IP
13:42.13znhIs it possible to force asterisk to connect to it's local IP?
13:42.44Guggegive the phone an internal ip, and set it to connect to the asterisk internal ip
13:42.52*** join/#asterisk dmz (n=dmz@64.203.233.195.dyn-cm-pool35.hargray.net)
13:43.13znhIt has an internal IP. Asterisk is router's NAT
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13:48.57DJCharlieMorning all. Anyone awake?
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13:55.03trentsterHey have setup asterisk on a intel nano platform, if at compile time I choose all the audio greetings in all the formats like g729 etc will asterisk automatically choose the correct version and not transcode greetings etc/
14:01.18DJCharlieI'm wondering why it won't execute a script using the System() function....
14:04.23znhGugge: ?
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14:35.46DJCharlieOk folks. Why isn't this working?
14:35.57DJCharlieexten => 301,5,System(/mnt/vm/incoming/xt/vmsend.sh "record301" "user@mydomain.com")
14:36.20DJCharlieThe script runs perfectly if I execute it by hand.
14:44.06Guggemy guess it a path problem with the script
14:44.40Guggetry calling the script with an empty PATH variable, and see if it works
14:46.02DJCharlieNope, the paths are absolute in the script, and it's running as the same user as asterisk.
14:47.20Guggeset verbose to something high, and check what happens on the asterisk console
14:48.19DJCharlieGood idea. Setting it to 10.
14:49.10*** join/#asterisk Sandheaver (n=jeremiah@98.214.112.102)
14:50.03DJCharlieNothing.
14:50.16DJCharliePlays the goodbye message, and hangs up.
14:52.04DJCharlie-- Executing [301@xt301:5] System("Zap/1-1", "/mnt/vm/incoming/xt/vmsend.sh") in new stack
14:54.24kaldemarCLI won't give any output when you run something with System
14:54.45kaldemarhow is it not working? is the script not receiving the arguments?
14:55.56DJCharliekaldemar: No, I've even gone so far as to try hard coding the variables into the script for testing. It never actually executes.
14:57.05DJCharlieRunning the script by hand, it works perfectly.
14:58.33kaldemardoes the script have execute permissions?
14:58.55DJCharlieYes.
14:59.17kaldemaryou should be seeing the arguments in the "Executing..." line.
14:59.43DJCharlieFor testing, I have temporarily hard-coded the args.
15:00.09kaldemaris the script calling a binary?
15:00.40DJCharlieyes.
15:00.42kaldemarhave you tried to echo $1 and $2 to a file to verify that it actually executes?
15:00.53kaldemaris the script calling the binary with full path?
15:01.01DJCharlieYes. And it doesn't output anything.
15:02.13kaldemarthis doesn't do a thing: http://pastebin.ca/1580300 ?
15:02.37DJCharlieThat's similar to what I have in my script, and the answer is no.
15:03.53kaldemarwell, there's obviously something wrong with it. if you're willing to pastebin "ls -la /mnt/vm/incoming/xt/vmsend.sh" and the script contents, i can take a look if i spot something.
15:06.28DJCharliehttp://pastebin.com/d25a8405b
15:06.55DJCharlieThat's with the script hard-coded
15:08.17kaldemarare you running asterisk as root?
15:08.24DJCharlieYes.
15:09.31radenSandheaver: morning
15:09.38Sandheavermorning
15:09.40*** join/#asterisk misteranonymous (n=tyler@udp115943uds.hawaiiantel.net)
15:09.42radenguess what
15:10.15radenI picked up a procurve 1800-24G for my desk for $125 on ebay last night
15:10.30radenthe guy has 2 left if you need any
15:10.31Sandheavernice
15:10.41Sandheaverlink?
15:10.55Sandheaverthat a PoE switch?
15:11.17Sandheaverdid you try virtualization?
15:12.46radenSandheaver: no didnt try yet gf wants to go out of town for weekend :(
15:12.59Sandheaverfair enough
15:13.05SandheaverGF > computers
15:13.12radeni want to thought
15:13.24Sandheaverunless it's just a getting to know you better weekend, those usually suck
15:13.49DJCharlieEnjoy those while you can. After the wedding, everything changes.
15:13.49kaldemarDJCharlie: i can't see a why the script wouldn't execute, if all you've said is correct.
15:13.55Sandheaverit doesn't take long, but if i were you i would be thinking about the GF
15:14.13radenGF = PITA
15:14.19DJCharliekaldemar: I can't either. I'm not exactly new to scripting.
15:14.40radenDJCharlie: i dont plan on getting married we have a kid together thats enough
15:15.22radenSandheaver: plus all monitors at work ATM trying to get everything done , you get the linky i offered 125
15:15.47Sandheaveryup got link
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15:17.09radencheap for a L2, 24 port, gigabyte w/ lifetime warranty
15:25.15DJCharlieOkay, NOW the echo statements in the script show the script is running.
15:25.26DJCharlieBUT, nothing is actually happening.
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15:31.29misteranonymousTimToady_: there more than one way to do it
15:34.58znhI have a phone connected to a router. The Asterisk server is on the LAN behind the router. If I try to dial my echo service, I can't hear anything. I did rtp debug and it showed the external IP. Is this what it should be?
15:40.26sircolinmy asterisk box is not local to me, when I do the same rtp debug it shows my external ip too so im guessing that is correct
15:50.05p3nguinznh: Maybe you need to check out the NAT workaround.
15:50.24p3nguinBut then again, maybe not.
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16:45.57DJCharlieOk folks, I'm back. System is working great, but I'm thinking towards future expansion...
16:46.28DJCharlieNow, correct me if I'm wrong, but an FXS is what allows you to plug in a standard analog telephone, right?
16:46.47kaldemarright
16:47.31DJCharlieOkay. Say I need 3 phones, all 3 set to ring when a certain extension is dialed. I'd need a 3 port FXS?
16:48.31kaldemaror 3 FXS ports, yes. however, if you're thinking about getting phones, i'd go for VoIP ones.
16:49.34DJCharlieAnd those are rather expensive... I'm working on an extremely shoestring budget here.
16:49.48*** join/#asterisk nightrid3r (n=kvirc@78.20.232.172)
16:50.25DJCharlieOr would 3 VoIP phones be cheaper?
16:50.39kaldemarproper fxs cards aren't cheap either.
16:50.59*** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk)
16:51.08DJCharlieWell, my FXO card is a X1000P, but it works fine. :)
16:52.03*** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk)
16:52.41hardwiremoo
16:53.00DJCharlieSo which would be cheaper?
16:53.01*** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk)
16:53.30kaldemarhttp://store.digium.com/productview.php?product_code=1TDM440EF <-- for that price, you can get 4-5 decent VoIP phones and you'll have a different grade of reliability and scalability.
16:53.32[TK]D-FenderDJCharlie: Do those phones have to act independantly and only ring together on this one occasion?
16:53.55*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
16:54.22DJCharlieWhile that'd be nice, not really. It's perfectly fine if they all work together 24/7.
16:54.34*** join/#asterisk imcdona (n=t@c-24-22-222-115.hsd1.wa.comcast.net)
16:54.42[TK]D-FenderDJCharlie: I mean you could plug 3 phones onto 1 ATA's jack
16:54.48DJCharlieThey'll be incoming calls only, and rarely used at that.
16:55.24[TK]D-FenderDJCharlie: Otherwise just get yourself 2 x Linksys PAP2T-NA's for < $100 total
16:55.42DJCharlieOk, [TK]D-Fender, you've lost me. Is an ATA the same as the FXS?
16:55.57[TK]D-Fender~ATA
16:55.58infobotfrom memory, ata is Analogue Terminal Adapter which provides an FXS and/or FXO and ethernet, see http://www.voip-info.org/wiki/view/ATA
16:56.41DJCharlieOkay, so I get one of those, put a 3 way splitter on it, and plug in the 3 phones, right?
16:57.22nightrid3rDJCharlie: doesn't that defeat the whole idea of using a pbx
16:57.47DJCharlienightrid3r: We're mainly using it for voicemail to be emailed to our DJs.
16:57.57nightrid3roh k
16:58.10DJCharlieThe only calls that would ring those phones is station emergencies.
16:58.55[TK]D-FenderDJCharlie: You could get away with just 1 then
16:59.22DJCharlieSo what's a good low-cost brand/model for that?
16:59.23[TK]D-FenderDJCharlie: they'd be like having 3 phones in your house ont he same line.  So anyone picking up the phone can listen in on the others
16:59.33[TK]D-FenderDJCharlie: I just gave you a specific model
16:59.36DJCharlieThat's not an issue. :)
16:59.45[TK]D-FenderDJCharlie: http://www.telephonydepot.com/Catalog/Cisco-Analog-Adapters/Linksys-PAP2T-NA
17:02.02DJCharlieThat's a nice price. :) Just have to convince the Bosslady.
17:05.00DJCharlieOk, I'm off to a staff meeting to squeeze more money from the turnip.
17:05.05DJCharlieThanks folks!
17:11.14*** join/#asterisk luca`gervasi (n=ashura@host147-165-dynamic.55-79-r.retail.telecomitalia.it)
17:11.17luca`gervasiHallo
17:14.57Dovidhello
17:19.19fofwarehello guys
17:19.53Dovidhallo
17:22.04Sandheaverhillo
17:22.20Sandheaverhollo
17:22.22Sandheaverhullo
17:22.28Sandheaverhyllo
17:22.34Sandheaverall vowels covered
17:24.23Sandheaversomeone should make some sort of adapter that you can plug your cell phone into when you get home, so that when your cell phone rings all the phones in the house ring, and you can just pick up any phone and talk via your cell phone
17:24.28Sandheaverthough i suppose call forwarding could do this
17:24.57Sandheaverbut i'm talking about a completely standalone phone system at home, that is useless for anything other than intercom when a cell phone isn't providing service to the system
17:25.18Sandheaverguess it doesn't make a lot of sense if you use your phone for anything other than talking though
17:29.05fofwareSandheaver: take a look http://www.google.com/googlevoice/about.html
17:29.21*** join/#asterisk haryv (i=lanny@174.1.114.16)
17:29.48Sandheaveryeah i have a google voice account, but i want to make calls out of the home phone system, via the cell phone as well
17:30.15Sandheaveri seem to recall a product like this for nokia phones, using nokia's popport
17:32.23*** part/#asterisk Orbixx (i=Orbixx@office.exoware.net)
17:33.40[TK]D-FenderSandheaver: This device already exists
17:33.48Sandheavergasps
17:33.52[TK]D-FenderSandheaver: lets you use your cell as an FXO
17:33.57Sandheaverreally
17:34.00Sandheavernice
17:34.14Sandheaverwhat's it called
17:34.21[TK]D-Fenderhttp://www.phonelabs.com/prd05.asp
17:34.57Sandheaveroh bluetooth
17:35.04Sandheaverwhy the F didn't i think of that
17:35.10Sandheaverawesome
17:35.11drmessanoloves chan_mobile in Asterisk
17:35.21SandheaverooOOoohh
17:35.22drmessanoOOPS, sorry, that was on topic
17:35.26Sandheaverhah
17:36.14drmessanochan_mobile works great and costs me existing_system+$3 dongle
17:37.08fofware[TK]D-Fender: Hello, I did changed mailcmd in voicemail.conf for my own script, and now I need monitoring and debug it, can you tellme how I can do that?
17:40.09[TK]D-Fenderfofware: What is there to monitor?  Is your script not getting called?
17:42.18fofware[TK]D-Fender: look like it's not called, for that I want to get some error or something to find the issue
17:42.18p3nguinWhen using an IAX trunk, calls do not go through.  I see the following message on the console: chan_iax2.c:3253 __auto_congest: Auto-congesting call due to slow response.  Where do I need to start in order to solve this?
17:42.47fofware[TK]D-Fender: when i put mailcmd=/usr/share/asterisk/asteriskmail
17:42.58fofware[TK]D-Fender: nothing happend
17:44.15fofware[TK]D-Fender: the script supose write in file the parameters that receive, so after I can write my own mail-script like you told me
17:47.15[TK]D-Fenderfofware: And where do I see you leaving a VM, and your config showing me that your script should get called? or seeing the permissions ont he script itself?
17:52.29haryvis there a test script that can run though my asterisk system to locate problems with it? Seems i get random complaints such as, pressing one extention and the user experiances dead air. I can press it though the ivr and it will work fine.
17:52.37*** join/#asterisk scalex000 (n=chatzill@206puntacana02.codetel.net.do)
17:53.07fofware[TK]D-Fender: http://pastebin.ca/1580465
17:58.00fofware[TK]D-Fender: script is working I test it from command line and work, so have permissions and everything that need
17:59.32[TK]D-Fenderfofware: you have not shown me HALF of what I ask for and I will not take your word for it that the users match
18:03.20fofware[TK]D-Fender: sorry I don't understand, I can paste voicemail.conf if you want and anything that you need, only let me know
18:04.27[TK]D-Fenderfofware:  I want to see the LS of the script.  voicemail.conf CLI output of you LEAVING a VM
18:06.24fofware[TK]D-Fender: Ok, what you mena with LS of script?
18:06.55[TK]D-Fenderfofware: "ls -la /usr/share/asterisk/"
18:07.08fofwareok
18:08.43fofware[TK]D-Fender: voicemail.conf http://pastebin.ca/1580481
18:10.30fofware[TK]D-Fender: ls http://pastebin.ca/1580484
18:12.34fofware[TK]D-Fender: CLI output http://pastebin.ca/1580486
18:15.23fofware[TK]D-Fender: in system.log after leave VM each time i see
18:15.23fofwareSep 26 15:14:02 shila asterisk[9718]: rc_avpair_new: unknown attribute 1490026597
18:15.45*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:15.49fofwareuse or not the script
18:16.22[TK]D-Fenderfofware: did you restart * completely after changing voicemail.conf?
18:17.05fofware[TK]D-Fender: yes, I did try both, reload from CLI and stop and start from command line
18:17.17*** join/#asterisk ruben23 (n=RPL@122.55.48.243)
18:17.45*** join/#asterisk linuxviewer (n=example@ip68-110-115-3.ph.ph.cox.net)
18:17.54fofware[TK]D-Fender: Asterisk is 1.4 over debian, only to let you know
18:18.48fofware[TK]D-Fender: SMTP  postfix, pop3 and IMAP Dovecot
18:19.03[TK]D-Fenderfofware: go prove the user * is running as
18:20.16fofware[TK]D-Fender: ?, in other world, please...
18:21.24linuxviewerMy asterisk server is no longer sending voicemail notifications via email.  Is there a place that has the logs for email to diagnose why it is no longer sending email notifications?
18:21.36luca`gervasidoes anybody use Cisco 7940 phones with asterisk?
18:22.28linuxviewerluca'gervasi - Yes
18:22.49[TK]D-FenderLinuCheck your mailer
18:23.01[TK]D-Fenderlinuxviewer: Check your mailer
18:23.24fofwarelinuxviewer: your mail server can send mail to same mail-box that you have in asterisk?
18:24.22*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
18:24.41fofwarelinuxviewer: remember from some months ago all mail providers reject mails from unknow mailservers
18:25.00linuxviewerfofware - I do not understand your question.  It has been working, and then out of no where, I no longer receive the emails with the voicemail.s  Can I just try by typing "mail" in CLI and put in information?
18:25.32linuxviewerfofware - I am aware that this may be the case, so I am trying to figure out if the machine is actually sending the email, or if it is running into an error of some sort (of course, if it shows it is sending the mail, but no mail is received, then we know it is a provider problem)
18:26.07linuxviewerI assume by default, asterisk uses mail not sendmail?
18:26.11fofwarelinuxviewer: I'm verry new in asterisk, but that I did is, check your mail server
18:26.38linuxviewerfofware - you say to check mail server, but I am unsure what you mean by that.  How do I go about checking mail server?
18:26.52fofwarelinuxviewer: what mail server that you have installed
18:26.59linuxviewerUses sendmail if not mistaken,
18:27.02[TK]D-Fenderlinuxviewer: The sample configs all call sendmail.  If your install is using that script for something else, well you should already know where to look.  otherwise its sendmail
18:27.12linuxviewer./var/logs/maillog/ shows that it is sending the email, but it is not received.
18:27.32linuxviewerSo possibly ISP blocking the outbound email?
18:27.51p3nguinUse the mail relay they provide for you.
18:27.52fofwarelinuxviewer: Yes, that is the cause
18:28.08p3nguinoften relay.yourisp.net or similar
18:28.35linuxviewerp3nguin - I see.  I put that relay information in sendmail configuration file somewhere?
18:28.41p3nguinyeah
18:28.51fofwarelinuxviewer: yeah
18:29.21linuxviewerThank you for your assistance everyone.
18:29.36linuxviewerI will contact ISP, get their relay information, put it in /etc/mail/sendmail or google where it needs to go.
18:29.49fofwareno problem linuxviewer
18:30.15[TK]D-FenderLinutons of ISPs prevent you from sending mail.  You may have to configure sendmail to use their outbound relay with auth, etc.
18:30.26[TK]D-Fenderlinuxviewer: Rather common
18:31.07fofwarelinuxviewer: lastnight I see in internet some configuration of Asterisk voicemail to use one account of gmail, maybe you can find it
18:31.46fofwarelinuxviewer: I did lost the link but is there in some place :o)
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18:35.47e4Are parentheses allowed in the username field in iax.conf?
18:36.03[TK]D-Fendere4: Don't
18:36.35e4[TK]D-Fender:  It's the setting that Teliax gave us :/
18:38.34e4It seems at the very best like a bad idea to set that and it's not connecting properly, but I wondered if it's an invalid setting entirely.
18:39.55*** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net)
18:39.59e4I'd be open to suggestions for a better host if that's the issue.
18:41.10[TK]D-Fendere4: You haven't shown us the problem.
18:41.22[TK]D-Fendere4: you took 1 blind stab and have no backup in hand
18:43.19*** join/#asterisk denon (i=denon@sassinak.net)
18:43.19*** mode/#asterisk [+o denon] by ChanServ
18:45.03fofware[TK]D-Fender: so what you mean with "go prove the user * is running as"
18:47.55*** join/#asterisk [8none1] (n=[8none1]@c-68-52-23-77.hsd1.tn.comcast.net)
18:48.49*** join/#asterisk e4 (n=e4@76.79.48.214)
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18:50.08e4[TK}D-Fender:  iax2 show registry shows it registered to the provider, but when I try to call into the service I get 'chan_iax2.c:10507 socket_process: ... Registration refused' from the provider.
18:52.04fofware[TK]D-Fender: I did find the issue, for some reazon, when i call lua script from asterisk do not execute it, maybe don't read the interpreter
18:52.32fofware[TK]D-Fender: I did write other in Shell and it Work
18:52.53fofware[TK]D-Fender: thanks
18:55.31fofware[TK]D-Fender: Asterisk export to shell VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM, VM_CIDNAME, VM_DATE ?
18:55.55box2what the balls
18:56.18box2dpkg-reconfigure tzdata sets the new timezone in /etc/timezone and /etc/localtime
18:56.30box2but when i echo $TZ it still says my old timezone
18:56.38box2and when i use date it also still shows my old timezone
18:57.13box2i hate everything
18:57.54[TK]D-Fenderbox2: .... wrong channel
18:58.08[TK]D-Fenderbox2: Unless you want to abstractly state that you hate * as well
18:58.17[TK]D-Fenderbox2: at which point the door is on your left :)
18:59.55p3nguinWhen using an IAX trunk, calls do not go through.  I see the following message on the console: chan_iax2.c:3253 __auto_congest: Auto-congesting call due to slow response.  Where do I need to start in order to solve this?  Debug output from the time I dial the number until I end the call: http://pastebin.ca/1580538
19:18.47box2[TK]D-Fender: heh
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20:04.38cilkayHello. I'm new to all this VOIP stuff but not new to telephony. In the '90s, I did quite a bit of development using Dialogic analog and digital phone cards using their C toolkit. I'm going to order some DIDs. Is the caller ID that is displayed on outgoing calls a feature of the DID provider or something I configure at my end from Asterisk?
20:05.08cilkayIf it's at my end, can that ID be changed on a per call basis?
20:05.32[TK]D-Fendercilkay: * can tell the provider what to show, but they have to allow it
20:05.43[TK]D-Fendercilkay: Just as many allow it as not
20:06.03[TK]D-Fendercilkay: You'll have to ask.  Tryincally those who offer more bulk-like termination support this
20:06.14[TK]D-Fendersilespecially the per-minute providers
20:06.42cilkayWhat would be an example of one or a couple that do?
20:07.08luca`gervasii need a flat to call italian numbers and a local (+39) phone number...do you know any?
20:07.52p3nguincilkay: VoIP.ms and Vitelity both allow passing of your own CID.
20:08.27cilkayp3nguin: I was on the voip.ms page, ironically.
20:08.40p3nguinCallCentric does not.  They require you to verify you have control over the number and then you can select it from the web interface.
20:09.22p3nguincilkay: I use VoIP.ms for my home and office termination since my call volume is low.  At 1 cent per minute, it's hard to beat.
20:10.23cilkayMy use case: I'm associated with multiple companies. When I return or make calls from numbers associated with those companies, I want the names and phone numbers of those companies to show.
20:10.42cilkay1 cent per minute is close enough to free that it doesn't make much difference at low volumes.
20:10.53p3nguinYou'll be able to set the CID number, but the name is done in another way.
20:10.55cilkayAnd probably not at high volumes either.
20:11.13*** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk)
20:12.05cilkayp3nguin: How is the name set then?
20:12.19p3nguinCNAM lookups are done on the receiving end.
20:13.01cilkayAnd the name is associated with the number by the ITSP?
20:13.25p3nguinWhen you call Customer1, you only send the CID number.  Customer1's telco does a lookup of the number in their LIDB.  If there is a name associated with it, the name will display on the caller ID unit.
20:13.26cilkayIf so, that would be fine, as long as I can set that name once and forget about it.
20:14.02p3nguinFor example, I might call you, and your CID box will say Illinois Call rather than my name.
20:14.10cilkayWhere does the name get associated with the number?
20:14.22p3nguinin the telco's database(s)
20:15.03p3nguinThink of it like a reverse DNS lookup.
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20:15.19cilkaySo, if your company is Penguin Consulting, is there a way for you to specify to your ITSP that you want that name displayed when my telco does a CNAM lookup when you call me?
20:15.47cilkayI understand the analogy. I'm still not clear who does that association.
20:16.09*** part/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk)
20:16.18*** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk)
20:16.18p3nguinSometimes the ITSP can submit your CNAM (Caller id NAMe) to a large database, but there is nothing to guarantee that the called phone company uses that db.
20:17.16*** join/#asterisk sircolin (n=sircolin@my83-216-68-241.mynow.co.uk)
20:17.17cilkayThat doesn't seem very robust.
20:17.53p3nguinOn the other hand, if you are using a phone number of a company that has a regular land line number already, they most likely have a CNAM already in place.  If you dial out using that phone number, the name lookup will be successful and will show the name of the company.
20:18.48p3nguinMaybe I can give you a working example.
20:19.18p3nguinIf you receive a call from your local city hall, your caller ID will display something like CITY HALL plus their number, right?
20:19.32cilkaycorrect
20:19.47p3nguinTheir number has the name already associated with it, so any lookup will be successful.
20:20.11p3nguinIf I set my CID number to that of your local city hall and then I call you, you will see CITY HALL and their numbre on your caller ID.
20:20.29p3nguinMake sense?
20:21.08cilkaySure, but what if you set your CID number to the actual VOIP number you got from voip.ms?
20:21.18p3nguinI send their number.  Your telco does a name lookup based on that number, it succeeds.
20:21.24cilkaySay it was a previously unallocated number.
20:21.52p3nguinIf the number does not have any name in the telco's database, it will show something like OUT OF AREA plus your number.
20:22.09p3nguinMight say ILLINOIS CALL like when I call you from my IL number.
20:22.25p3nguinYou've probably seen out-of-state calls on your ID before.
20:22.42cilkay... which to many people signals "telemarketer".
20:22.54p3nguinUnfortunately, yes.
20:23.58p3nguinYou can always ask your ITSP to enter your name in the database, and then be very hopeful that it will eventually propagate to the db that your callee's telco uses.
20:24.16cilkayMaybe the "trick" it to associate with a POTS number that is "aged", i.e. has a name associated with it.
20:24.58p3nguinIf they have a name in their db, that's what will show up when you use it to call someone.
20:25.07p3nguineven if it's not your name.  :)
20:25.19cilkayYou'd have to port that number to your ITSP eventually. It's a more expensive way to go but it's probably more reliable.
20:26.16p3nguinEven porting some existing number to a VoIP service isn't a guarantee that the name will persist in the previous telco's database.
20:26.37p3nguinThey could drop it once they release the number.
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20:26.47cilkaytrue
20:26.55p3nguinI don't know if it's normal for them to drop them or not, though.
20:27.03*** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com)
20:27.04cilkaySo I'll just have to impersonate "IBM" then. :)
20:27.14p3nguinThat's the easy way.
20:27.31cilkay"Hello. I'm calling from the White House."
20:27.38p3nguinyeah  :)
20:28.05fofwareI'm in Pink House, hello cilkay
20:28.07p3nguinThe problem is that they won't have a callback number on their cid for you, though.
20:28.54cilkayfofware: Hello. I'm in a light tan brick house but I have white trim :)
20:29.18_trineI'm in the dog house according to my wife
20:30.57drmessano_trine: Lucky.  My wife added a new level called the "outhouse" and it makes the doghouse look like a stay at the Ritz
20:31.11_trinehe he he
20:31.36fofwaredrmessano: lol
20:32.29cilkayp3nguin: CNAM Caller name Lookup - 1.25¢ per query <== from https://www.voip.ms/specifications.php I'm not sure what that means.
20:32.49_trinewell I got my asterisk sending me my voicemail nicely
20:33.01drmessanoI knew I was in the outhouse when I woke up on wednesday and had a crescent moon drawn on my chest with a sharpie
20:33.08drmessano:(
20:34.07_trineasterisk works very well off my linksys router
20:35.17_trineanyone that runs asterisk off a computer is a big girls blouse :P
20:35.43drmessanoAnyone who runs Asterisk off a router doesn't have many friends
20:36.05_trinethese north west of Englands sayings always make me laugh
20:36.46_trinedrmessano: theres not many of us yet
20:37.20drmessano_trine: Yet?  What you're doing is not new.. Asterisk has been run on Linksys boxes for at least 4 years now
20:37.36drmessanoIts just not a great idea in general
20:37.41_trineyes but not with everything included
20:37.44drmessanoLittle horsepower, limited feature set
20:37.58_trineI have everything loaded
20:38.20drmessanoYou have g729, chan_skype, chan_mobile with a bluetooth dongle, and you're using ODBC logging for CDR?
20:38.22_trineand I have only used 3% of my flash space
20:39.05_trineI still have 97% of my flash space unused
20:39.14_trineon my router
20:39.29*** join/#asterisk parker (n=ti@189.24.120.122)
20:39.37parkerboa tarde
20:42.22parkerestou tento um problema com o asterisk 1.4.21.1
20:42.22parkertenho uma fila de atendimento, a fila funciona normalmente com os ramais internos, porem quando a ligação é externa e vai para fila, a mesma so tem duração de 1:18 min
20:42.22drmessanoWhat exactly is "everything"?
20:42.23parkertanto do LINK E1 quanto do tronco SIP Vono
20:42.23_trineparker: we all have problems
20:42.23_trinedrmessano: everything I can find
20:42.23parker_trine: my  problem is with queue trougth external trunks
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20:43.15parker_trine: they only have a time of 1:18 min, aswered or not
20:43.15drmessano_trine: Considering there are some things I KNOW will not run on that platform, what is "everything"?
20:43.15_trinedrmessano: routers if you are a single user as I am are now more than equal to the task
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20:43.16_trinedrmessano: as i said I have only used 3% of my 8 gigs of flash space on my router
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20:43.24drmessanoYou're not explaining much of the everything part
20:43.35parker_trine: any suggestion?
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20:43.46drmessanoLike for example, how is meetme working out for ya?
20:44.10_trinedrmessano: everything I have found for asterisk I have found I have loaded into my router
20:44.31drmessanoFrom the Openwrt repo?
20:44.43_trineparker: sorry I have no suggestions for you but maybe one of the other guys could help
20:44.59parker_trine: tanks
20:45.04parker_trine: thanks
20:45.47_trinedrmessano: from everywhere,,  but of course that means there will be more to add but so long as it does not take more than 8 gigs of space I'll be ok
20:46.28p3nguincilkay: Those are lookups on your end when someone calls you.  They have to query that database I was telling you about.  Without paying per query, when someone calls you, you'll only get the CID number on your display.
20:46.34drmessano_trine: Some of that stuff is not even working.. so just loading a bunch of random crap doesnt imply any real functionality
20:46.52_trinedrmessano: which stuff isn't working
20:47.39parkercan someone help me with my queue problem? (asterisk 1.4.21.1)
20:48.02cilkayp3nguin: That could get more expensive than the phone company's overpriced visual caller ID if you get a lot of incoming calls.
20:48.24p3nguincilkay: It's the difference between "Caller ID" and "Caller ID with name" services.
20:48.43cilkayWhat my monopolist telco calls "Visual Caller ID".
20:48.50p3nguinah
20:49.05cilkayActually, even at 1.25 cents per lookup, it's still a bargain by comparison.
20:49.06p3nguinAT&T calls it Caller ID with name.
20:49.34_trinedrmessano: I also have freeswitch loaded on my router too
20:49.46cilkayI think my rip-off telco charges $15/month just for that feature.
20:50.32p3nguinI think if you wanted to by the 'with name' part a la carte from the telco, it's probably around $5 per month for unlimited queries.
20:50.39p3nguinhere, I mean
20:51.04p3nguinBut most plans include the name lookups in the rate.
21:01.11*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
21:02.28*** join/#asterisk ChannelZ (i=channelz@burner.com)
21:03.43ChannelZAnyone using Skype for Asterisk?
21:07.11p3nguiniax2-provision.c:518 iax_provision_reload: No IAX provisioning configuration found, IAX provisioning disabled.   What exactly do I need to configure to provision it?
21:08.07ChannelZiaxprov.conf
21:12.42p3nguinThe IAX provisioning is required for a trunk to the provider?
21:12.49ChannelZnot necessarily.
21:14.04ChannelZprovisioning is like automatic configuration
21:14.17p3nguinI have termination service and I currently use SIP for that.  I have SIP phones on the inside of the Asterisk box.  I wanted to change from SIP to IAX2 for the termination, but keep the phones as they are.
21:15.14ChannelZthat's fine.  You just need to get IAX info from your service provider and set that up
21:15.38ChannelZand then change your dialplan so it dials out on the right channels etc
21:15.51p3nguinI got the config info and put it all in.  After I switch over to IAX, calls don't complete.  Here's a debug from the time I dial the number until the time I hangup: http://pastebin.ca/1580538
21:16.20p3nguinWhen I reload iax2, it spat out that notice about no provisioning.
21:16.30retentiveboyI'm looking at the Polycom ACD login/logout features and am wondering if there's any support for it in * 1.6.  Anyone using it successfully?  Or, should I just make a login/out dial-plan extension and a speed-dial?
21:16.47p3nguiniax2 show peers does show my ITSP peer.
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21:18.25ChannelZIt looks like a communications error, like the traffic isn't getting through.  Is your Asterisk box a or behind a firewall?
21:18.52p3nguinThe asterisk box is on a public IP address and it does have iptables on it.
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21:19.28ChannelZis port 4569 open?
21:19.31p3nguinAll outbound traffic is allowed, and the iax port is also open for inbound (although I don't know if that's important for outbound calls).
21:19.36p3nguinyeah
21:20.02e4Finally have everything else hooked up.  I'm getting 'CallToken Support required' on incoming calls even though the setting to disable the new security measures is set as per the pdf release.  Is anyone else had issues with it?
21:20.26ChannelZhmmm
21:20.51ChannelZdo you get incoming calls?
21:21.12p3nguinAsking me or e4?
21:21.18ChannelZyou, sorry
21:21.41ChannelZnot sure what e4 is talking about, sounds like you had to have been here earlier for the first part of the story :)
21:22.06p3nguinI have a SIP trunk from another provider for incoming calls.  This IAX configuration will be only for outbound calling.  Using SIP on the same provider, the calls go out normally.
21:22.17*** join/#asterisk manxpower (n=EWIELING@24.42.221.26)
21:23.32p3nguinI just wanted to tinker with IAX in place of SIP.
21:25.56p3nguinThe problem, as you may have noticed in the call debug was  chan_iax2.c:3253 __auto_congest: Auto-congesting call due to slow response.
21:26.05ChannelZyeah
21:26.28ChannelZwhich seems like a block of communication in one direction or another; whether completely or massive packet loss or what
21:26.34p3nguinFrom the looks of it, since I don't have iax devices, I don't need to worry with provisioning.
21:26.56ChannelZright that provisioning warning is fine, I get it too and I setup an IAX connection between home and work
21:27.20ChannelZWhat 'type' is your account in iax.conf ?  Peer, friend..
21:27.24p3nguinI really expected it would be like SIP -- create a peer, create a dialplan that is compatible, make calls.
21:27.42p3nguinfriend
21:28.28ChannelZit pretty much is like SIP
21:29.10ChannelZSo all UDP traffic out is allowed, and incoming UDP 4569 is allowed?
21:29.19p3nguincorrect
21:30.26ChannelZhmm
21:35.21p3nguinThe packet count on port 4569 increases when I try to make a call, so something's getting back from the provider.
21:35.35ChannelZif you only have outgoing call support should your type be 'peer' and not a friend?
21:36.03ChannelZI'm not sure why that would matter but..
21:36.08p3nguinI don't know that it affects it, but I am willing to change for testing purposes.
21:36.56p3nguinsame problem
21:37.36ChannelZso you hear nothing and then just get disconnected?
21:38.11p3nguinTakes about 2-3 seconds, then I get a fast busy (congestion).  I have to manually end the call.
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21:41.48p3nguinAccording to that debug, it's 5 seconds, actually.  :)
21:42.36ChannelZwell 4?  4001ms
21:43.03ChannelZAll I can think is something is not setup right on the provider side fully
21:43.18p3nguinI'll try several of their other servers, then.
21:43.30p3nguinThey run Asterisk one some and OpenSER on some.
21:43.31ChannelZyou said the packet count on 4569 increases when you make a call, on the INPUT table I'm assuming?
21:43.38p3nguinyes
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21:44.10ChannelZhuh.  Stumped
21:45.49p3nguinSame thing on all four US servers.
21:45.51ChannelZI haven't quite used IAX like you, I've got a softphone using IAX into my * server
21:47.47ChannelZwhat does your entry in iax.conf look like
21:49.47p3nguinhttp://pastebin.ca/1580691
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21:57.46ChannelZI really dunno.  Seems like a problem on their end, if it's showing up as registered you've got communication going both ways
21:57.51ChannelZwhat version of Asterisk are you running?
21:58.03p3nguin1.4.24.1
22:01.50ChannelZdo you have voicemail through them or any other 'local' extension you can dial other than an outside number?
22:06.37loubenpaw
22:07.36p3nguinYeah, I can set up their echo test extension real quick.
22:09.54p3nguinEven calling those internal numbers gives me that auto-congestion message and a fast busy.
22:20.51*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
22:20.55fofwarehey guys, May I define a varible in voicemail.conf and send it to my customized mail script?
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22:33.29luca`gervasihow can i setup a ring group?
22:34.59ChannelZas in ringing multiple extensions?
22:35.50luca`gervasiyes
22:36.32ChannelZjust glob them together with & in your Dial command.  Like Dial(SIP/joe&SIP/bob&SIP/gina)
22:37.34manxpower~answers
22:37.35infoboti heard answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
22:40.54luca`gervasithanks
22:40.55luca`gervasibye bye
22:46.53ChannelZargh this damn skypeforasterisk is making me mad
22:55.30fofwareChannelZ: do you know if May I define a varible in voicemail.conf and send it to my customized mail script?
22:57.00manxpowerfofware: I doubt that you can.
22:57.33manxpowerbut if you read up on the docs of what variables ARE passed to the mailscript you might be able to set one of those to the value you want
22:57.52fofwaremanxpower: thanks verry much, maybe that the cause I have 2 days trying and nothing :o)
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22:58.46manxpoweror you can save the value of the variable in AstDB or a file on the disk or something like that and have your script read the value
22:58.58fofwaremanxpower: I guess I tryed that too, I will try againg, thanks
23:01.00fofwaremanxpower: other way is read the value from table of SQL but I want evite 2 reads thanks
23:01.55ChannelZwhat is the data/variable you're wanting to define?  Something unique to each mailbox or what would be the use of configuring something in the .conf file
23:03.07fofwareChannelZ: I want send mail in spanish to spanish people in english to english poeople and ....
23:03.42fofwareChannelZ: so I have a context of eachone
23:03.47ChannelZah
23:04.18fofwareChannelZ: but mailbody only accept setting in general context
23:04.49*** join/#asterisk Ad-Hoc (n=nimbus@ppp209-68.adsl.forthnet.gr)
23:05.07ChannelZso you need to write a completely different mailbody
23:05.08Ad-Hochi ppl
23:06.17fofwareChannelZ: the only I need send some value to an extenal script to know if laguage is english/french/spanish/portugues/etc
23:06.41fofwarebut I don't find the way to do that
23:07.03ChannelZyeah built-in I don't think there is a way
23:07.36ChannelZyou'd have to just make your 'wrapper' program read the contents of the mail and parse it and do something else
23:07.47*** join/#asterisk Tim_Toady (n=moi@adsl163-162.kln.forthnet.gr)
23:07.59fofwarethe other problem I found is my script done before asterisk voce mail done to send data
23:08.19ChannelZhuh?
23:08.22fofwareChannelZ: yes
23:08.53manxpowerI suspect the VM context is passed to the script.
23:09.00[TK]D-FenderOr do the easy thing and mod something you CAN send to endoe that data
23:09.05fofwareChannelZ: no problem with the time, script can wait
23:09.34[TK]D-FenderBut of course that would require some actual creativity... and we wouldn't want that, now would we?
23:09.46[TK]D-Fenderis off for a while
23:11.37ChannelZthe mailcmd is assumed to be a delivery agent, like sendmail - my understanding is the code is constructing the entire mail (including the attachment if that's turned on) and that's just being fed to the 'mailcmd' over stdin
23:11.50NovceGuruanybody know if vitelity charges for toll free termination?
23:11.53ChannelZso even if you could put a variable on the commandline being called, it wouldn't necessarily help you
23:15.26fofwareChannelZ: yes it send all by stdin and by default to sendmail, but you can change it to other script, and read stdin and make customized mail
23:15.57ChannelZright but you have to parse that mail to replace the bits you want and keep the rest
23:16.29fofwarebut look like in voicemail process is not possible add any data
23:17.53ChannelZShort of patching the code to add that capability, the "easier" way might be to just construct the emailsubject and emailbody in such a way that you can parse it easily in your wrapper script, and use the VM_MAILBOX as a switch to choose what language your script does
23:17.58ChannelZlike:
23:18.37ChannelZemailsubject=[MSGNUM:${VM_MSGNUM}][MAILBOX:${VM_MAILBOX}]
23:18.49*** join/#asterisk WinZ (n=winz@82.146.61.218)
23:19.46WinZhello guys, anyone using asterisk with net2phone here? what are the settings?
23:19.47fofware[TK]D-Fender: My english is too poor, but that you did say sond like you don't like I make questions in the channel, If i'm wrong I'm sorry, If not thanks anyway
23:19.53ChannelZyour script can then search/parse for something looking like [MSGNUM:*][MAILBOX:*] and pull that out
23:20.09fofwareChannelZ: yes I have all data
23:20.23fofwareinclusive the file attached
23:21.17ChannelZso then parse the mailbox number in your script and use that to determine what language to reconstruct the message in
23:21.53fofwareChannelZ: but I did to modify one of that vars or add another one but i can't do that
23:22.14ChannelZin splits your configuration to be in two places (voicemail.conf for the actual mailboxes and your script for the language) but again, short of writing a bunch of new support into the actual asterisk voicemail code to support an extra variable passed on the commandline....
23:22.35manxpower"The way it works is basically any time that somebody leaves a voicemail on the system (regardless of mailbox number), the command specified for externnotify is run with the arguments (in this order): context, extension, and number of voicemails in that mailbox. These arguments are passed to the program that you set in the externnotify variable. "
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23:26.38manxpowerYou could also encode the information in the e-mail address i.e.  fr^timmy@lassie.com then in side your script remove the fr^ from the e-mail address before sending the message
23:26.49ChannelZexternnotify would work but then you have to write a bunch more code to read the voicemail it's self and encode it to attach to an email
23:27.10fofwarethanks manxpower and ChannelZ I will make another file to read langue value of the estensions
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23:27.38ChannelZyeah manxpower that would be easiest actually
23:29.29fofwareyes, that is a good point, manxpower but I don't want do that because is not for my own instalation
23:29.54manxpowerfofware: there is no easy way to do what you want.  There are only not-easy not-pretty ways.
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23:33.44fofwaremanxpower: thanks verry much, I'm only looking for one way that don't interfere with standart configurations files
23:34.29ChannelZwhat are you writing the script in
23:34.45fofwareso I guess one good thinks can be read with script the language of extension from other conf files
23:35.31fofwarei'm writing in shell and lua, because I make web interfaces to routers
23:38.06fofwareand both scripts work very well
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23:45.59WinZ!seen aurax
23:46.29*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
23:55.54*** join/#asterisk phunyguy (n=phunyguy@h69-130-65-171.kgldga.dsl.dynamic.tds.net)

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