IRC log for #asterisk on 20111023

00:01.11The-KernelNGT's broadsoft isn't going to let me change something like that
00:01.32dijibp3nguin, what mail clinet would your dialplan use to send email? mutt?
00:03.31*** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net)
00:03.43p3nguinYes, mutt.
00:03.57p3nguinAnd I use ssmtp or msmtp as my MTA.
00:04.12p3nguinDid use msmtp, now us ssmtp.
00:04.21p3nguins/us /use /
00:15.04*** join/#asterisk adeel (~adeel@24-246-63-106.cable.teksavvy.com)
00:37.09dijibi think my mutt line is ok now... i was just about to test some changes,... and im trying to use some voicemail authentication thing but not sure if its working as i was always getting authenticated.
00:39.37p3nguinIf you are being authenticated, that sounds like it works.
00:56.34dijibhttp://pastebin.com/qrKN0kRW doesnt work
00:57.26p3nguinline 6 is bad.
00:57.59p3nguinI still don't understand why you insist on using SayAlpha when there's SayDigits.
00:58.23p3nguinLine 9 has pipe lines instead of commas.
01:00.09dijibcaught that
01:00.15dijibline six is what?
01:00.18dijibplayback?
01:00.21p3nguinThen you try to email a file that hasn't even been created, yet.
01:00.32dijibok so do that after?
01:00.53p3nguinYou'll need to do the emailing in the h extension, after the call has ended.
01:00.54dijibor dial? see i dont know how to climax
01:01.16dijibso goto(h,1)
01:01.18dijib?
01:01.19p3nguinno
01:01.37p3nguinCalls go to extension h when they hang up.  That's what h means.
01:01.40p3nguinh is the hangup extension.
01:02.00p3nguinexten => h,1,System()
01:02.15p3nguinor similar
01:02.19dijibin the same context?
01:02.29p3nguinIt will run h in the current context.
01:02.36dijibsee im getting a bit confused with the contexts and questions how i could add security..
01:02.58dijibgo ,Hangup(); goes to h?
01:03.02dijibin that context?
01:03.13p3nguinWhen a call hangs up, it goes to h.
01:03.37p3nguinThe Hangup() probably will never actually get used, but I always define it anyway.
01:03.42p3nguinlike a safety net.
01:04.38dijibright now i have h defined as such since you defined it. man that needs a cheque
01:04.39dijibexten => h,1,Goto(h-${FAXSTATUS},1);
01:05.08p3nguinNot to mention, you have a subject of "new fax from..." to email something that isn't a fax.
01:05.14dijiblol
01:05.19dijibthats minor
01:05.27p3nguinDid you ever read the book?
01:05.31p3nguinMy guess is no.
01:05.36dijibi wrote the book
01:05.46dijibjust not this book
01:05.46p3nguinNow I know you're lying.
01:08.21dijibcurrent dialplan is 529 lines
01:08.30p3nguinI re-did my fax extension, too, just so you know.
01:08.49dijibwell im thinking send h- to email
01:09.19dijibbut am i right by trying to still use the CALLERID(num) variable in h-?
01:09.38dijibor even using h-
01:09.39p3nguinExtension h is where thing happen after the call ends.  You're trying to email a file of a recording of a call that hasn't yet been made.
01:10.07dijibright but with how its set right now, if there is no fax it send the call to h-
01:10.31p3nguinYou're not even dealing with a fax.
01:10.43p3nguinDon't expect a fax setup to work when not dealing with fax.
01:13.46dijibthen in the main context define h?
01:15.21p3nguinI don't know what a "main context" is to be able to answer that appropriately.
01:16.50carrar[main]
01:17.30dijibinternal
01:17.44dijib{}
01:17.46dijib][
01:23.38*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
01:25.36*** join/#asterisk addeswe (~adde@c-65bce255.013-16-756d651.cust.bredbandsbolaget.se)
01:41.27p3nguinThis is how I email a fax, or lack of a fax, if that's what happened:  http://pastebin.com/fbSwh0hd
01:41.58p3nguinThe same concept can be used to email something that wasn't a fax.
01:49.55p3nguinThis is probably how I would do what you were trying to do:  http://pastebin.com/Zr0HvK00
01:51.11p3nguinBut I can't expect you to use the dial plan I write for you.  That would be too easy.
01:54.38p3nguinYou could even create DB entries for every phone you have with a unique caller ID number, and assign an email address to it, then you could email the recorded file to your own email address when it finishes.  I'll edit the paste to reflect this idea.
01:55.48p3nguinDone./
01:57.37*** join/#asterisk zerohalo (~zerohalo@74.60.136.128)
02:03.57dijib-- ${DB(fax/fax-manager/email)}); is causing me issues. ive got them store like this
02:04.09dijibemail 9055551324 email@domain.com
02:05.16dijiboh but im going it like this sorry
02:05.54dijib-- ${DB(email/${CALLERID(num)})});
02:07.04dijibi cant do that can i.
02:09.44p3nguinYou've got a lot of }) in there.  I think you have an extra set.
02:10.03p3nguinIf you have a database entry for email/9055551324 with a valid email address, it should work.
02:10.45p3nguinNah, you have the right amount of brackets.
02:11.00p3nguinIt just appears like a lot without seeing the rest of the line.
02:16.19dijibits still causing APPERROR in status
02:16.42dijibill paste what im using now.
02:16.55dijibbut its basically your code with my System line
02:20.38p3nguinI'm sure it's your mailing part that is not working.
02:23.30p3nguinAre you using the sudo portion like I use?
02:27.01p3nguinI'm going to fall asleep waiting on you to paste a single line.
02:30.33*** join/#asterisk autofsckk (~autofsckk@unaffiliated/autofsckk)
02:38.39dijibi think so
02:38.42dijibyes im using sudo
02:38.45dijib-u asterisk
02:39.52p3nguinDid you configure sudoers for that to work?
02:39.58dijibhttp://pastebin.com/fRd8JcXZ
02:40.02dijibsudoers?
02:40.04dijibmaybe not
02:40.44p3nguinThat's... not what I wrote for you.
02:40.52p3nguin(2049.54) <p3nguin> This is probably how I would do what you were trying to do:  http://pastebin.com/Zr0HvK00
02:41.23p3nguinAnd you have to configure sudoers to do that.  It's not magic, so it won't work without being configured.
02:41.55p3nguinvisudo
02:42.01p3nguinadd the following line:
02:42.03p3nguinasterisk ALL=(ALL) NOPASSWD: /usr/bin/mutt
02:42.23p3nguinsave, exit, try again.
02:42.54p3nguin(2051.10) <p3nguin> But I can't expect you to use the dial plan I write for you.  That would be too easy.
02:51.56*** join/#asterisk gxdssoft (~gxdssoft@190.236.111.135)
03:12.23*** join/#asterisk Devon_ (~chatzilla@63.214.236.169)
03:16.34leifmadsenassuming the US border agency decides not to be douchy, see you all at AstriCon!
03:21.09*** join/#asterisk ChrisInSydney (~Chris@60-242-81-231.tpgi.com.au)
03:40.11*** join/#asterisk LiuYan (~LiuYan@222.125.132.191)
03:40.39*** join/#asterisk lovetide (~LiuYan@222.125.132.191)
03:46.02*** join/#asterisk Rufus (Rufus@unaffiliated/rufus)
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04:15.33*** join/#asterisk BuenGenio (~Gene@n219076175228.netvigator.com)
04:40.10dijibanybody still alive?
04:40.50WIMPyNo
04:40.56WIMPyThis is a Zombie Channel
04:41.03dijibthought so.
04:41.10dijibi wish there were intelligent zombies
04:42.18WIMPyJust add some fresh blood.
04:43.47lovetide:D
04:43.59*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
04:57.44dijibfor the life of me i cant get this working
04:57.46dijibsame  => n,System(/bin/echo "Please see attachment."|/usr/bin/sudo -u asterisk /usr/bin/mutt -a "${MIXMONITOR_FILENAME}" -s "Recording from ${CALLERID(num)}" -- ${DB(email/${CALLERID(num)})});
04:59.21*** join/#asterisk WiretapSeven (~Wiretap@unaffiliated/wiretap)
05:00.45SeRidijib, You trying to sent a vmail to an email with attch?
05:01.04dijibsomething like that yeah
05:01.06WIMPysudo -u asterisk?
05:01.24dijibyes.
05:01.25WIMPyAsterisk doesn't run as astersik, but mutt should?
05:01.37dijibasterisk runs as asterisk
05:01.46dijibsafe_asterisk runs as root
05:01.54WIMPyThen why do you dudo from asterisk to asterisk?
05:02.22dijibi dont know. im impotent. ask p3nguin
05:02.49WIMPyAnd the command to be executed by sudo isn't clear. It's neither quoted nor are the options terminated by --.
05:03.38dijibquotes where
05:03.43WIMPyIs that a cpuy&paste gone wrong thing?
05:03.53dijibcopypasta?
05:04.02SeRiwhy not use "su userid -c "command"
05:04.32WIMPyIt only makes sent when changing user, doesn't it?
05:04.45WIMPys/sent/sense/
05:07.08*** join/#asterisk irroot (~irroot@41-135-56-110.dsl.mweb.co.za)
05:08.14*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
05:08.16SeRidijib, when I want to send emails with attch inside my linux using asterisk I usually keep it away from the dial plan to eliminate over head in the dial plan.
05:08.27SeRihere is an example of my email out script
05:09.00SeRihttp://pastebin.com/7nfsm9vh
05:09.24SeRiits a script that runs in the background and looks for the tiff file in tmp
05:09.31SeRithan converts it to pdf and emails it
05:09.51SeRiIs not what you want to do but it gives you an idea how to use scripts to email out
05:10.23dijibit working now
05:10.23[TK]D-Fenderdijib: You've haven't shown us that dialplan failing
05:11.49dijibi have it emailing out now.
05:11.59dijibbut i need to fix my gotoif script
05:12.10kaldemardijib: do you still want the voicemail e-mail address in dialplan?
05:12.42dijibhere ill show you what i have right now.
05:14.53kaldemar${CUT(AST_CONFIG(voicemail.conf,${CUT(SIPPEER(mailbox),@,2)},${CUT(SIPPEER(mailbox),@,1)}),",",3)}
05:15.30dijibhttp://pastebin.com/vvegExVx
05:16.36SeRiI would have a script monitoring for the file created and email it out and than ether back it up or remove it
05:17.13[TK]D-Fender$[["${result}" = "0"]?
05:17.16[TK]D-Fender2x [
05:17.17p3nguinOr you could just let asterisk email the file when the call hangs up.
05:17.19SeRiThats just me though :)
05:17.52dijibthats an error?
05:17.53SeRito me that just creating a bunch of overhead in the dial plan but hey I know nothing...
05:18.01[TK]D-FenderDiyou have [[ in a row
05:18.30p3nguinIt's not any more overhead than a daemon monitoring for new files.
05:18.30dijibk
05:18.45*** join/#asterisk leftist (~leftist@50-10-91-159.gar.clearwire-wmx.net)
05:18.46[TK]D-Fenderkaldemar: the e-mail in voicemail is voicemail.conf, not sippeer
05:18.56[TK]D-Fenderkaldemar: that is the VM box & context
05:19.00[TK]D-Fenderkaldemar: not at all the same
05:20.35p3nguinHaving a small amount of work to do after the call hangs up isn't really a big deal, since it's a computer built within the 20 years.
05:20.46p3nguinthe last 20 years, that is.
05:21.19p3nguinYou're creating more overhead doing the actual recording than you are using mutt to email the file when it is done.
05:21.21SeRip3nguin, like I said that's the way I do it. I like to keep things a bit simpler and separate from each other. dial plans can get extensive and complicate so to kee it simple for me I separate the two.
05:22.30kaldemar[TK]D-Fender: my example had errors in the SIPPEER calls, but it gets the email from voicemail.conf using AST_CONFIG.
05:23.01kaldemar[TK]D-Fender: the VM context and box are just to get the right line from voicemail.conf.
05:23.07SeRiby the way I have not been able to go to the mail. school and work is killing me
05:23.16SeRiare*
05:23.33p3nguinAny idea about that other SIM?
05:23.58SeRiyes those are coming this coming week.
05:24.21kaldemar${CUT(AST_CONFIG(voicemail.conf,${CUT(SIPPEER(${CHANNEL(peername)},mailbox),@,2)},${CUT(SIPPEER(${CHANNEL(peername)},mailbox),@,1)}),",",3)}
05:24.27kaldemar^ that really works.
05:24.32SeRiI am hoping to see the guy at school on tuesday.
05:25.55[TK]D-Fenderkaldemar: Yeah, I over-focussed on the inner call...
05:25.59[TK]D-FenderI am tired...
05:26.17dijibwhats wrong with my AGI line?
05:26.22dijibwhy doesnt it like me?
05:26.30[TK]D-Fenderdijib: Who said anythine was wrong with your AGI line?
05:26.34dijibafter i took the [ out im getting invalid
05:26.37[TK]D-Fenderdijib: You haven't shown us the failure
05:26.55dijibim getting a permission denied on the agi script
05:27.11kaldemar[TK]D-Fender: it happens to the best of us. except to me ofcourse. :)
05:27.41[TK]D-Fenderkaldemar: You're clearly excluded from the "best of us" Vvenn group ;)
05:29.11dijibfailier: http://pastebin.com/vcjRmsCv
05:29.33p3nguinPermission denied.  Seems clear to me.
05:29.58[TK]D-FenderAs it said.. permissions issues... and You aren't showing any real details up front and I don't have time to wring them out.
05:30.08dijib4 -rwxr-xr-x.  1 asterisk asterisk 1447 Oct 22 17:47 chk_vm_pwd.agi
05:30.08[TK]D-FenderI'm off...
05:30.59p3nguinHow about the directory above that?
05:31.20WIMPyAnd what does the first line say?
05:31.29p3nguinI usually use namei to see the perms on the entire path.
05:36.18SeRip3nguin, I wanted to mention that comcast did found an issue with port 5060
05:36.29p3nguinThey admit to blocking it?
05:36.46SeRino. they politically chose there words
05:36.57p3nguinI figured.
05:37.19p3nguin"We have found an anomaly."
05:37.36SeRiMr.Blah we have found a discrepancy in our configuration been used in your modem. we have addressed and we will credit your account accordingly.
05:38.18p3nguinDo they give you credit per day per port?
05:38.38SeRinah they just credited me for 1 month.
05:39.01p3nguinIf they are going to base it per port, divide whatever you think you should get by 65536.
05:39.10SeRirofl!
05:39.41*** join/#asterisk seanbright (~sean@asterisk/contributor-and-bug-marshal/seanbright)
05:39.43p3nguinThat's what I'd do if I were Comcast.
05:40.12p3nguinI'd be the manager in charge of the people that are hired to piss off customers.
05:40.12SeRilol
05:40.15*** join/#asterisk kaushal (~kaushal@59.161.24.233)
05:40.19kaushalHi
05:40.46kaushalasteriks uses only one core out of 8 cores on the server, any known issue ?
05:41.15p3nguinI'd train them with material taken from my own experiences.  I rarely have a satisfactory transaction with any company.  Ever.
05:41.30SeRip3nguin, funny part is that they would not release any details. to this day I have yet to receive the managers call that was suppose to explain the issues to me.
05:41.45p3nguinIs the problem fixed?
05:41.48SeRiYes
05:41.55p3nguinThat's the important part.
05:41.55SeRiI am using port 5060 :)
05:42.01SeRiYes sr it is.
05:42.21p3nguinDidn't you say you used to have phone service with Comcast?
05:42.27SeRiYes.
05:42.32p3nguinThat's why it was jacked up.
05:42.43SeRiMaybe but they claim the dont use sip
05:42.46p3nguinThey never changed your config after you canceled the voice service.
05:43.02SeRithey use there own proprietary protocols an shit
05:43.13p3nguinI'd bet they don't.
05:43.29*** join/#asterisk BuenGenio (~Gene@n058152141152.netvigator.com)
05:43.30WIMPyOr just an IAD eating everything that arrives on 5060?
05:45.40p3nguinIf you look at the protocols that your voice-enabled DOCSIS modem supports, you'll probably find that it's nothing special and certainly nothing proprietary.
05:46.21SeRiI try and it does not mention anything
05:46.31MDesadeim back ladies...
05:46.42*** join/#asterisk lovetide (~LiuYan@222.125.132.191)
05:46.43MDesadewhats the latest greatest?
05:47.06p3nguinRepresentatives of service providers are trained to lie to customers, and not trained in the actual field of services being provided.
05:47.34SeRiThats true
05:47.41p3nguinAsk any cable company employee that you reach when you call about anything DOCSIS related, and they don't have a clue.
05:47.44kaushalWIMPy: hi
05:49.16WIMPyMDesade: Doing you mapping now?
05:50.41SeRiVoice over Internet Protocol (VoIP) is a technology used to transmit voice and related calls over a data network.  Most VoIP service providers use the public Internet to transmit your calls.  Comcast does not; we use this technology to transmit your calls over our advanced broadband network.
05:50.46SeRiThat cracked me up
05:51.06MDesadei tried asteriskNOW, but did not install... doesn't support the LSI RAID card in that version of CentOS... so, i installed CentOS-6-64bit and am reading about the differences between CentOS and Kubuntu
05:51.32WIMPySeRi: MAybe they use two IP networks. My provider even uses 4.
05:51.41p3nguinThat statement is probably entirely accurate.
05:52.13dijibthis took me ages
05:52.13dijibhttp://pastebin.com/nbWW0wMq
05:52.18SeRip3nguin, it might be but if they ware using voip I am sure they are using standard protocols.
05:52.41*** join/#asterisk FainaUkraina (~Gene@203.145.92.197)
05:52.41p3nguinThey aren't doing it over public internet, they are doing it over their cable network, which is using RFC 1918 addressing.
05:52.46SeRiWIMPy, could be. I really dont care. I have my port back :) yay...
05:53.04p3nguinThey are using VoIP, and I could almost guarantee you that they are using a standard protocol.
05:53.41p3nguindijib: namei -l /var/lib/asterisk/agi-bin/chk_vm_pwd.agi
05:54.03p3nguinOr if you don't have -l, maybe you have -m and -o at least.
05:54.07WIMPyYes, most of them use SIP. That's why their lines all suck big times.
05:54.32SeRilike you said the important thing is that it works. after some battle but it now works :P
05:54.41p3nguinexample:  http://pastebin.com/qMcjFzHb
05:54.48dijibhttp://pastebin.com/JPJRWWRJ
05:54.56SeRicomcast just wants to much money for there piece of crap service.
05:55.19SeRiI was paying 39.99 a month for there voip
05:55.28SeRiI am down to 8 dollars a month
05:56.30p3nguinOkay, so it shouldn't be a problem with the perms on the agi file.  How about the perms on the interpreter?  (That's found in the shebang in the agi script.)
05:57.00p3nguin$8 per month?  For what?!
05:57.04SeRihttp://maps.level3.com/default/
05:57.17SeRivoipms
05:57.51p3nguinOh, I thought you meant $8 per month for internet service from Comcrap.
05:57.57SeRirofl
05:57.59SeRihell no
05:58.01SeRiI wish!
05:58.22dijibhttp://pastebin.com/NFhp1rLT
05:58.22p3nguinI had no idea they'd even print out your bill for only $8.
05:58.49p3nguinCan asterisk execute /usr/bin/php?
05:58.54dijibno
05:59.00p3nguinThat's a problem.
05:59.02dijibactually thats the problem
05:59.18p3nguinYou forgot to install php?
05:59.24dijibshould i install php or use the non php script
05:59.42kaldemaror just not use agi in the first place?
05:59.43dijibyum install php5 ?
05:59.44p3nguinIf you can do it without php, do it without php.
05:59.49dijibok
06:00.10*** join/#asterisk Marquel (~Marquel@static.132.171.47.78.clients.your-server.de)
06:00.18Marquelmorning.
06:00.58dijibother script
06:00.59dijibhttp://pastebin.com/wa6Y7JF3
06:01.15SeRip3nguin, lol. fam does not use the phone much. Just me for work and the office. most calls are inside voipms network so they are free :)
06:01.49p3nguinIf you're going to use that one, don't forget to install perl.
06:01.58Marqueli have a little problem with asterisk-1.8.7.1: it tells me signalling=bri_net_ptmp is not implemented for chan_dahdi. but asterisk-1.8.6.0 does support this signalling mode. what's wrong with that?
06:02.08dijibwhich would you use perl or php?
06:02.16p3nguinBut if you can do it without an AGI, do it without an AGI.  See: kaldemar.
06:02.55p3nguinI'd choose no AGI first.  Then I'd choose perl second.
06:03.08p3nguinThen python.  Then php last.
06:03.27dijib${CUT(AST_CONFIG(voicemail.conf,${CUT(SIPPEER(${CHANNEL(peername)},mailbox),@,2)},${CUT(SIPPEER(${CHANNEL(peername)},mailbox),@,1)}),",",3)}
06:04.00dijibhe said SIPPEER was voicemail.conf ?
06:04.29p3nguinSIPPEER() can check certain values associated with your SIP peer.
06:04.41p3nguinIn this case, mailbox is the value it is looking for.
06:05.15p3nguinIt uses AST_CONFIG() to find the matching value in voicemail.conf.
06:05.17dijibso mailbox is ${CALLERID(num)} ?
06:05.41p3nguinUsually your mailbox number is going to be the same as your caller ID number which is usually the same as your extension number.
06:05.44kaldemardijib: no, mailbox is what you have configured for a peer in sip.conf in the mailbox parameter.
06:05.46p3nguinBut it really doesn't matter.
06:05.54p3nguinBecause it is checking the mailbox setting for the peer.
06:05.58p3nguinNothing to do with caller ID.
06:06.06p3nguinIt is channel/peer based.
06:06.47dijibwhat if there is no peer?
06:06.51p3nguinThere is.
06:07.03p3nguinWithout a peer, there is no phone call.
06:07.06dijibits a call comming from itsp
06:07.12p3nguinThe ITSP is a peer.
06:07.29dijibso peername = callerid(num) ?
06:07.32dijibno.
06:07.34p3nguinNot usually.
06:07.48p3nguinBut I'm not sure what your ITSP has to do with your voicemail.conf.
06:08.22dijibok nothing. i wanted the caller to enter their CID number. then use that to check against voicemail.conf's account
06:08.25dijibs
06:08.44dijibext_num ext_psw
06:08.47dijibpwd
06:08.48p3nguinThey are defining their own caller ID number to make a call?
06:08.57dijibyes
06:08.57kaldemardijib: is the number entered by the caller also the mailbox name in voicemail.conf?
06:09.07dijibyes
06:09.09kaldemardijib: what is the context for the mailboxes?
06:09.30p3nguinI'm interested to see where this goes.
06:09.35dijib600 => 123,Hugo Chavez,chrismfinn@gmail.com
06:09.38dijiboi.
06:09.41p3nguinThat's not a context.
06:09.42dijibsanitised.
06:09.48dijiboh default.
06:10.13kaldemardijib: what variable are you using to read the caller-entered value?
06:10.27dijibext_num
06:10.38dijiband ext_pwd for passcode
06:11.12kaldemar${CUT(AST_CONFIG(voicemail.conf,default,${ext_num}),",",3)} gets you the e-mail address then.
06:12.52dijibive got mutt getting the email from ${DB}
06:13.03dijibi need to authenticate users against voicemail.conf
06:14.43kaldemarheh, now i actually know what you're trying to do.. oh well, Authenticate(${CUT(AST_CONFIG(voicemail.conf,default,${ext_num}),",",1)})
06:15.09dijibi guess i dont express myself very well
06:16.22kaldemarand the same thing is done by VMAuthenticate(${ext_num})
06:16.23kaldemarguess so.
06:16.24dijibi just wanted someone to call in. use the dialplan set CID & authenticate at the same time. enter destination number, call out, record call and email the recorded file
06:16.55dijibim thinking of using your cut instead of the DB as i would need to add users in DB and if i use CUT i can just use voicemail.conf
06:17.37kaldemaror use VMAuthenticate.
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06:21.28Marquelnobody?
06:22.21WIMPyMarquel: How old is your libpri?
06:24.28irrootSMH there plenty ref leaks in app_queue
06:24.47MarquelWIMPy: quite a bit - 1.4.12, no newer available from distribution.
06:25.19WIMPyMarquel: That's good enough.
06:25.35WIMPyDid you make Asterisk yourself?
06:25.48MarquelWIMPy: gentoo-linux, yes, compiling myself.
06:26.36WIMPyThen something must have gone wrong. Did you install that version of libpri before compiling Asterisk?
06:27.32Marquelyes. did not recompile/reinstall libpri since july 2011, yet asterisk 1.8.7.1 does not support bri_net_ptmp, 1.8.6.0 does.
06:29.21WIMPyAh, so you already had it working. NFI then. Maybe some hickup in the current version?
06:29.30Marquelasterisk's configuration shows libpri as usable.
06:29.37Marqueldunno, that's why i ask.
06:29.53Marqueli'm about to file a bug with my distribution.
06:30.44Marquel(i can at probably will also email my hardware vendor about it, maybe they do have an idea)
06:30.49Marquels/at/and/
06:31.06WIMPyBTW.. irroot: Did you put your chan_lcr patch online somewhere whe I can link to it?
06:31.52WIMPywhere
06:37.20MarquelWIMPy: so, bug filed, now goes teh email to teh hardware vendor...
06:38.02WIMPyJust out of interest: What hardware are you using?
06:38.44MarquelWIMPy: a junghanns.net duoBRI PCIe dual-channel isdn adapter.
06:39.39WIMPyInteresting to often see duo bris in use.
06:40.00Marquel;)
06:40.04WIMPyNo space for two cards?
06:40.10MarquelWIMPy: no.
06:40.56Marquelthe mainboard of that box has exactly one PCIe slot (last one of those) and i do not intend to buy (expensive) cards which i can not use with the next mainboard.
06:41.19WIMPyUSB :-)
06:41.23Marqueland why two cards when one of them can perfectly handle inside ISDN and outside ISDN?
06:41.28Marquelerm... no.
06:42.16WIMPyI've never been a fan of USB, either, but it seems to work.
06:42.55Marquelwhatever can hide inside the box and be run fully on the box's powersupply does exactly that. i do not intend to clatter up my power consumption with inefficient supplies of some crappy usb-hardware or - also likely - consume all my USB-ports with just one energy-hungry appliance.
06:44.02WIMPyErr, you only need one usb port per port. No exta PSU, unless you need to power phones, but that's true for internal cards as well.
06:44.58WIMPyBut it would still be an interesting idea to maybe use two usb ports, like external HDDs do, and a step-up converter to supply power to a phone.
06:45.04Marquelyou are right with that ;)
06:45.14Marquelyou can't.
06:45.30WIMPyWhy?
06:46.06MarquelIIRC ISDN specifies 70 volts phantom power and not few amperes to do so. i like to see how you manage to do that with 500mA of 5V from USB ;)
06:46.52WIMPy2.5W. That's quite a lot.
06:47.19Marquelenough for one or two ISDN phones. not more.
06:47.53WIMPyDo you think you want to connect more than two phones to one USB adapter?
06:48.32Marquelit doesn't make much sense given that one phone can congest one of them by using both B-channels.
06:49.50Marquelbut ISDN specifies up to twelve phones on one S0-bus, and power-specs tell they should be all powered by that bus, if the bus is powered. ;)
06:50.05WIMPyYou shouldn't use more than one channel per phone, no matter how many calls you have.
06:50.30WIMPyNo, only 8 phones.
06:50.36WIMPy12 sockets.
06:50.45Marquelstill too much for 2.5W ;)
06:51.17WIMPyMaybe out of spec but should be good enough in practice.
06:52.38Marquelbut then, if you need a powered S0 - why not ask your provider for a set of NTSBs, configure them to be bus-slaves and hook them to the s0-bus and power-supply? ;)
06:53.25WIMPyThat's the usual way of doing it, but if you did it from USB you could be mobile.
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06:54.39WIMPyIt's just for fun anyway.
06:54.46Marquelmobile in "put my laptop with *, the usb-isdn-adapter and my isdn-phone in the hotel room, set up internet and vpn and use the isdn phone "from home" instead of paying hotel rates"? ;)
06:55.17WIMPySomethin like that :-0
06:55.18WIMPy)
06:56.07Marquelwell. i tend to just bring my headset and my laptop had a sip-softphone installed... somewhat less equipment. ;)
06:56.46WIMPyGood enough for your garden shed.
06:57.36Marquelbut then a friend of mine once came up with the idea of "hacking" one of the old analog phones with a number dialing wheel: apply a cell phone to the phone, have pulses converted to dial commands and a battery powerful enough to ring the electro-mechanical bell on incoming calls.
06:58.15Marquelwould be quite a show if that thing rings in public transport and you have to actually pickup a receiver connected to the phone with a cord. ;)
06:58.49WIMPyThat actually makes sense, as you would be allowed to use that while driving.
06:59.02WIMPyBut any commecrial GSM gateway would do that.
06:59.40Marquelyeah, but can you cramp a commercial gsm-gateway into those classical phones, along with such a large battery?
07:00.11Marquelbut guess how it'd look like if you decide to dial a number with that thing... ;)
07:00.31WIMPyYou can get both analog and ISDN to GSM gateways. And cars have lots of power.
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07:19.31*** join/#asterisk oej (~olle@ANice-257-1-31-202.w90-52.abo.wanadoo.fr)

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