| 00:00.08 | beek | I've started doing the calculations for the elimination of our proprietary system. It's costing us $9K/yr in maintenance costs and our magic number around here is two years... if you can pay off something in two years its worth it. I figure for $18K I should be able to add Asterisk boxen at each of our districts, replace the desksets and pay for it in two years. |
| 00:00.11 | hardwire | [TK]D-Fender: you hip to "deferred dialing" on CAS/E&M? |
| 00:00.36 | hardwire | I can't quite figure out what it's doing.. and if it's delaying dial time |
| 00:01.02 | *** join/#asterisk coppice (n=chatzill@46.166.17.210.dyn.pacific.net.hk) |
| 00:08.40 | *** join/#asterisk axisys (n=axisys@ip68-98-177-71.dc.dc.cox.net) |
| 00:10.33 | *** join/#asterisk myselfhimself (i=55ab28dc@gateway/web/ajax/mibbit.com/x-3e805952d4581433) |
| 00:10.36 | myselfhimself | hi |
| 00:10.45 | myselfhimself | can someone help opening SIP ports with iptables or other commands ? |
| 00:11.06 | myselfhimself | or just checking that those ports are open ? |
| 00:13.34 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 00:14.31 | bmoraca | Talkradio: this is a proliant DL380. there are no internal connectors of any kind. |
| 00:14.34 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 00:15.13 | [TK]D-Fender | bmoraca: Get drilling... |
| 00:15.42 | bmoraca | no kidding |
| 00:18.21 | *** part/#asterisk LemensTS (n=customgt@adsl-75-42-150-213.dsl.stlsmo.sbcglobal.net) |
| 00:22.28 | *** part/#asterisk SP_Garoto_Maroto (n=Yamato@unaffiliated/geek) |
| 00:25.33 | *** join/#asterisk Z_God (i=_ejabber@server1.jabbernet.eu) |
| 00:25.38 | *** join/#asterisk docid (n=eris@69.196.68.142) |
| 00:26.33 | docid | so, any ideas on what would cause hung up calls to leave their trunks open? as in i dial in from outside, dial out another line, and when i hang up both trunks are still on |
| 00:26.47 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 00:26.48 | Z_God | I'm having problems connecting a Twinke and Google Talk together |
| 00:26.57 | Z_God | both can dial the echo test fine |
| 00:27.19 | Z_God | but when I set up a call the audio only goes from twinkle to google talk, not the other way around |
| 00:27.44 | Z_God | it seems there is something wrong with the negatiation between the sip & gtalk channels |
| 00:28.03 | Z_God | what should I look for? |
| 00:32.29 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 00:33.14 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 00:33.28 | *** join/#asterisk intralanman (n=intralan@va-71-0-86-105.dyn.embarqhsd.net) |
| 00:37.35 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 00:40.23 | *** join/#asterisk Winkie (n=urmom@ur.fa.gs) |
| 00:40.37 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 00:42.15 | k-man | why would you want to connect asterisk to your jabber account? whats the benefit? |
| 00:42.43 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 00:44.38 | k-man | Z_God: what is Twinke? |
| 00:44.45 | Z_God | a SIP client |
| 00:45.24 | k-man | cool, its gpled too! |
| 00:45.39 | Z_God | yep, I think so |
| 00:45.44 | Z_God | http://www.twinklephone.com/ |
| 00:46.27 | voxter | god its ugly |
| 00:46.40 | msetim | There are somebody using 1.4.23? |
| 00:47.08 | Z_God | it works well though |
| 00:50.53 | Z_God | hmm |
| 00:51.06 | Z_God | phoning from twinkle to google talk works fine |
| 00:51.58 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 00:52.33 | Z_God | now asterisk is changing the call from alaw to ulaw |
| 00:52.46 | Z_God | it seems it needs to be processed by asterisk for it to work properly both ways |
| 00:52.57 | Z_God | is there a way to force that? |
| 00:53.15 | Z_God | the other way around I get a double ulaw call which doesn't work |
| 00:53.52 | Z_God | I'm using Asterisk 1.6 |
| 00:55.34 | *** join/#asterisk M4x51hah (n=M4X@190.18.148.4) |
| 00:55.52 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 00:56.13 | M4x51hah | i need some help configuring my VoIP carrier, a peer on sip.conf |
| 00:56.17 | M4x51hah | someone can help me ? |
| 00:56.31 | *** join/#asterisk outtolunc (n=me@c-71-198-197-201.hsd1.ca.comcast.net) |
| 00:56.34 | M4x51hah | im having quality of service issues |
| 00:57.27 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 01:00.24 | k-man | M4x51hah: whats the problem? |
| 01:02.03 | k-man | is asterisk 1.6 available in debian as a package? |
| 01:02.14 | Z_God | I made one myself |
| 01:02.35 | Z_God | there are scripts avaiable in a debian experimental svn repository |
| 01:02.58 | Z_God | I can make my packages public if you're interested |
| 01:03.32 | k-man | oh no, its ok |
| 01:03.41 | k-man | thanks - ill look into building it myself |
| 01:03.46 | hardwire | what is sf signaling? |
| 01:03.46 | k-man | Z_God: was it stable? |
| 01:03.58 | Z_God | it seems to work well |
| 01:04.04 | NovceGuru | Hm, I setup my telco line to call forward on busy to a sip channel, but when theres one call already fowarded and another comes in, it rings busy and I'm not out of channels |
| 01:04.08 | Z_God | other than the problem I have above |
| 01:04.14 | NovceGuru | is afraid what the telco will say |
| 01:04.15 | Z_God | it hasn't crashed a single time yet |
| 01:04.25 | NovceGuru | I know what they will say :( |
| 01:05.56 | *** join/#asterisk macros73_ (n=cs@c-71-61-74-104.hsd1.pa.comcast.net) |
| 01:10.02 | M4x51hah | i have a problem |
| 01:10.08 | M4x51hah | we have our asterisk 1.2 server on miami |
| 01:10.14 | M4x51hah | and our VoIP carrier ( peer ) |
| 01:10.19 | M4x51hah | that we use to perform SIP calls |
| 01:10.25 | *** join/#asterisk uluatu (n=uluatu@189.58.29.66.dynamic.adsl.gvt.net.br) |
| 01:10.25 | M4x51hah | to El Salvador |
| 01:10.32 | M4x51hah | is having audio issues |
| 01:10.55 | uluatu | anyone using 1.4.23.1 and has the updatecdr option set as "yes"? |
| 01:10.58 | M4x51hah | so basically our Asterisk Server has SIP peer on El Salvador |
| 01:11.00 | uluatu | it is not running for me. |
| 01:11.25 | M4x51hah | our Asterisk Server is located at Miami, and the peer is located on El Salvador |
| 01:11.37 | M4x51hah | for some reason, when i perform calls using that carrier |
| 01:11.52 | M4x51hah | there are some problems with the quality of the audio |
| 01:12.24 | M4x51hah | anyone have idea what can be going on ? |
| 01:12.32 | k-man | could it be latency or bandwidth issues between miami and el salvador? |
| 01:12.38 | k-man | or packet loss maybe? |
| 01:12.43 | M4x51hah | well, the pings are 60mbs |
| 01:12.46 | M4x51hah | we have another carrier |
| 01:12.51 | M4x51hah | that gives us pings of 260ms |
| 01:12.57 | M4x51hah | and that another carrier works well |
| 01:13.03 | M4x51hah | the other carrier is located in Argentina |
| 01:13.05 | M4x51hah | my question is... |
| 01:13.10 | msetim | uluatu: i'm |
| 01:13.20 | msetim | msetim: it's not working for me too |
| 01:13.21 | M4x51hah | our asterisk servers sends RTP packets from miami to el salvador via UDP ? |
| 01:13.41 | M4x51hah | the carrier has told us that it might be a packet loss issue |
| 01:13.51 | M4x51hah | we perform a traceroute |
| 01:14.07 | M4x51hah | and it takes 60ms |
| 01:14.12 | M4x51hah | same with pings |
| 01:14.17 | M4x51hah | but we dont get any packet loss on pings |
| 01:14.28 | M4x51hah | k-man: what do you think ? |
| 01:14.54 | M4x51hah | if its packet loss i need to see how i can perform the neccesary tests to confirm that the issue is really packet loss issues |
| 01:15.03 | k-man | M4x51hah: i'm not expert - but it could be something to do with packets arriving out of order maybe? I don't think ping could tell you if that was the case |
| 01:15.14 | M4x51hah | there isnt any way to switch the RTP protocol to use TCP instead of UDP so we avoid packet loss issues ? |
| 01:15.59 | rob0 | it would be worse! |
| 01:16.00 | k-man | M4x51hah: you might want to hunt around for more a more wrigorous (sp?) test of the network - and test it with a load also - ping might not send enough data to run into problems |
| 01:16.28 | M4x51hah | k-man: what is sp? SIP ? |
| 01:16.39 | M4x51hah | or sp is a command of the asterisk command line ? |
| 01:16.42 | k-man | spelling? |
| 01:16.51 | k-man | i couldn't work out how to spell wrigorous |
| 01:17.02 | rob0 | "Rigorous" |
| 01:17.09 | rob0 | close enough |
| 01:17.23 | k-man | dunno why, i thought it must start with w |
| 01:17.25 | k-man | bizzare |
| 01:17.38 | M4x51hah | how can i perform a more rigorous test of the network ? |
| 01:17.56 | *** join/#asterisk macli (n=macli@nmc.brc.ubc.ca) |
| 01:21.43 | k-man | M4x51hah: i'm not sure - but there care commercial programs that can do it |
| 01:21.51 | k-man | like http://www.tcpiq.com/ |
| 01:22.01 | k-man | s/care/are |
| 01:22.22 | k-man | hmmm |
| 01:22.41 | k-man | there is another tool that a friend uses - its client server and can do very thorough tests but I can't think of the name of it |
| 01:23.07 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 01:23.46 | *** join/#asterisk andresmujica (n=andresmu@ubuntu/member/andresmujica) |
| 01:23.59 | etfonhomey | ttcp |
| 01:24.17 | etfonhomey | You can have it do udp as well. |
| 01:25.14 | *** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-90cc0338329299d5) |
| 01:28.56 | k-man | rob0: why would sip over TCP be worse? |
| 01:29.58 | *** join/#asterisk mchou_ (n=mchou@unaffiliated/mchou) |
| 01:32.04 | *** join/#asterisk mchou (n=mchou@unaffiliated/mchou) |
| 01:32.53 | k-man | is it hard to implement billing in asterisk? |
| 01:37.44 | *** join/#asterisk minotaur01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 01:38.16 | M4x51hah | k-man: rob0 : SIP over TCP because of header overhead... however its not SIP that matters really, its RTP over TCP i think |
| 01:38.34 | M4x51hah | do you guys think that RTP over TCP is possible ? |
| 01:38.56 | k-man | M4x51hah: i have no idea, sorry |
| 01:39.20 | k-man | is there any asterisk conferences around anywhere? |
| 01:39.28 | k-man | appart from in the US? |
| 01:39.53 | minotaur01 | k-man: where are you looking for |
| 01:40.01 | k-man | europe somewhere would be nice |
| 01:40.04 | k-man | or in Australia |
| 01:40.26 | minotaur01 | don't know about europe |
| 01:40.34 | k-man | anywhere? |
| 01:43.51 | *** join/#asterisk maddog01 (n=test@dhcp-0-13-46-41-d9-29.cpe.mountaincable.net) |
| 01:44.18 | maddog01 | k-man: there is one in toronto in april |
| 01:44.54 | lmadsen | it360 |
| 01:44.59 | maddog01 | yup |
| 01:45.11 | k-man | maddog01: oh |
| 01:45.17 | k-man | id love to go to one actually |
| 01:45.24 | k-man | have you ever been maddog01 |
| 01:45.35 | lmadsen | has spoke at it for the last 3 years |
| 01:45.56 | *** join/#asterisk Kumbang (n=unknown@125.163.83.153) |
| 01:46.01 | maddog01 | no going to the toronto one for the first time |
| 01:47.51 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 01:48.30 | maddog01 | k-man: where is tehe site http://www.it360.ca/ |
| 01:49.48 | *** join/#asterisk TimRiker (i=timr@bzflag/projectlead/TimRiker) |
| 01:50.00 | TimRiker | ~1+1 |
| 01:50.01 | jbot | from memory, 1+1 is 3 for large values of 1, or 0 |
| 01:50.01 | jaytee | swear to god I'm getting Alzhiemer's |
| 01:50.46 | *** join/#asterisk joobie (n=joobie@mx01.anric.com.au) |
| 01:51.24 | joobie | hey guys.. i have an issue where 1 of 5 polycom phones keeps going unreachable and then reachable.. any methods i can use to try and troubleshoot this issue? |
| 01:55.36 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 01:55.39 | k-man | someone should do an asterisk conference in australia |
| 01:56.29 | *** part/#asterisk Z_God (i=_ejabber@server1.jabbernet.eu) |
| 01:59.10 | fatnasty1 | anyone have any experience configuring SMDI.conf? |
| 02:01.29 | M4x51hah | packet loss is the same thing that if the packets come out of order on UDP ? |
| 02:01.44 | M4x51hah | when the packets came out of order, then we say, packet loss exists ? |
| 02:02.45 | brunner | I am not able to get NVFaxDetect to compile. Would someone send me their compiled version of NVFaxDetect for 1.4? |
| 02:09.50 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
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| 02:22.41 | *** join/#asterisk Shaun2222 (n=shaun@ip68-5-154-128.oc.oc.cox.net) |
| 02:23.24 | Shaun2222 | ok, this isnt working for failover, basically it's dialing the same number 4 times if nobody answers during a timeout. |
| 02:23.25 | Shaun2222 | http://rafb.net/p/HnmGc672.html |
| 02:23.29 | Shaun2222 | what do i need to change here |
| 02:24.36 | brunner | if I originate a call with the Manager API, does it get hung up when I log off? |
| 02:28.35 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 02:31.17 | jaytee | brunner, don't believe it does since the call isn't going over the telnet connection, just the Originate command. |
| 02:33.41 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 02:33.41 | brunner | yeah, it doesn't. thanks. |
| 02:34.01 | *** join/#asterisk RobertLaptop (n=rmiddle@173-103-64-130.pools.spcsdns.net) |
| 02:35.53 | *** join/#asterisk RobertLaptop (n=rmiddle@173-103-64-130.pools.spcsdns.net) |
| 02:39.33 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 02:46.34 | brunner | does anyone have a throw-away voip account with any amount of credit I could buy from them? |
| 02:48.40 | *** join/#asterisk andresmujica1 (n=andresmu@ubuntu/member/andresmujica) |
| 02:49.30 | riddlebox | <PROTECTED> |
| 02:49.44 | brunner | is the activation quick? |
| 02:49.56 | keith4_ | voicepulse has a $2 credit, i think |
| 02:50.00 | keith4_ | "trial" account |
| 02:50.58 | k-man | anyone ever used one of these phones? http://www.zazz.com.au/index.php |
| 02:51.02 | k-man | are they any good? |
| 02:51.10 | k-man | open networks easychat 210 |
| 02:51.50 | keith4_ | ygwypf? |
| 02:52.42 | brunner | keith4_: nice, that's perfect. thanks. do they charge for out-bound toll-free calls? |
| 02:52.53 | k-man | keith4_: i guess so |
| 02:53.04 | keith4_ | i think all outbound calls are toll |
| 02:53.12 | keith4_ | not sure though. check their pricing page |
| 02:57.21 | *** join/#asterisk obnauticus (n=obnautic@about/windows/regular/obnauticus) |
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| 03:08.12 | drmessano | MagicBlaze007 is in #Freepbx trying to pretend he know something.. lovely |
| 03:08.35 | drmessano | Someone cant call between extensions ---> "probably a nat issue" |
| 03:08.41 | drmessano | Yep, thats my first thought |
| 03:10.48 | jaytee | drmessano |
| 03:27.19 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 03:27.38 | *** join/#asterisk Porks (n=Porks@unaffiliated/porks) |
| 03:27.39 | Porks | opa |
| 03:27.40 | *** part/#asterisk Porks (n=Porks@unaffiliated/porks) |
| 03:29.02 | jaytee | ok, I have absolutely no idea wtf that was all about. Anyone? |
| 03:30.21 | Qwell | jaytee: potatoes |
| 03:30.48 | jaytee | Qwell, huh? |
| 03:30.59 | Qwell | exactly |
| 03:31.03 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 03:31.03 | *** mode/#asterisk [+o file] by ChanServ |
| 03:31.11 | jaytee | haahaa |
| 03:34.24 | Qwell | looks at file and waits for the crash. |
| 03:34.59 | Qwell | turns his gaze to jaytee |
| 03:35.14 | jaytee | what? |
| 03:35.19 | Qwell | nothing :D |
| 03:35.26 | RichardLynch | I've set up Festival and can switch voices in festival.conf or wherever it was... I'd like to switch them on the fly. Is there some function I'm not finding in "the book" about this, or does anybody have a clue to donate? |
| 03:36.13 | jaytee | Colonel Mustard in the drawing room? |
| 03:39.09 | RichardLynch | I was thinking more along the lines of something useful to switch voices in Festival from a dialplan... |
| 03:39.28 | RichardLynch | with the candlestick |
| 03:44.28 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 03:44.29 | *** mode/#asterisk [+o file] by ChanServ |
| 03:51.34 | *** join/#asterisk file (n=file@asterisk/developer-and-muffin-lover/file) |
| 03:51.35 | *** mode/#asterisk [+o file] by ChanServ |
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| 03:59.04 | *** join/#asterisk codefreeze-lap (n=murf@72.21.67.40) |
| 04:00.18 | RichardLynch | I suppose I could re-define the tts_textasterisk function in festival.scm to take a third arg, the voice to use... I'd be a bit out of sync with the rest of the Festival users and the docs though. |
| 04:02.35 | RichardLynch | And I haven't hacked in Lisp for about 15 years, so am quite rusty... Tho I was pretty good at it once upon a time. |
| 04:03.24 | jaytee | I knew someone who was good at lisp but he finally went to a speech therapist and now he's much better. |
| 04:05.47 | prg3 | Can a queue with dynamic members, added by addqueuemember() have functionality like ackcall=yes allows with agents? |
| 04:06.46 | *** part/#asterisk RichardLynch (n=RichardL@c-98-193-37-55.hsd1.il.comcast.net) |
| 04:12.52 | [TK]D-Fender | prg3: AFAIK, no, but you can implement this yourself with the right kind of AQM |
| 04:12.56 | *** join/#asterisk Hanif08 (n=bucoo77@netop.jaring.my) |
| 04:21.41 | *** join/#asterisk CrashSys (n=kumba@azrael.crashsys.com) |
| 04:22.10 | prg3 | [TK]D-Fender: Got any more details on how that might work? |
| 04:22.36 | [TK]D-Fender | prg3: Local <- use some imagination about your DIAL |
| 04:23.49 | prg3 | [TK]D-Fender: Ah, the blatently obvious option.. play with dialplan for the outbounds. |
| 04:24.04 | [TK]D-Fender | ~[TK]D-Fender |
| 04:24.05 | jbot | [TK]D-Fender is the Zen Master of the blatantly obvious, and #asterisk 's resident Google-proxy. |
| 04:24.17 | [TK]D-Fender | :) |
| 04:24.25 | prg3 | Impressive :) |
| 04:24.26 | [TK]D-Fender | ~botsnack |
| 04:24.26 | jbot | aw, gee, [TK]D-Fender |
| 04:24.39 | [TK]D-Fender | ~areyouadog ? |
| 04:24.40 | jbot | Bark! Bark! |
| 04:24.45 | [TK]D-Fender | jbot: GOOD BOY! |
| 04:24.45 | jbot | [TK]D-Fender: :) |
| 04:24.59 | [TK]D-Fender | pets jbot |
| 04:25.41 | keith4_ | dear resident Google-proxy: what's 450 K in degrees F? |
| 04:26.28 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 04:26.32 | maxagaz | hi |
| 04:26.34 | keith4_ | you're a very slow google proxy. i even put it in a format that you could paste directly into google! |
| 04:26.52 | k-man | hehe |
| 04:29.20 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 04:29.22 | [TK]D-Fender | Googles.... prints the answer on 200 sheets of paper, rolls it up nice and tight and pulls a Jason Bourne on keith4 |
| 04:29.39 | [TK]D-Fender | whamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAM |
| 04:31.03 | keith4_ | alrightalrightalright, it's 350.33. I get it |
| 04:31.13 | Kobaz | details! |
| 04:33.51 | [TK]D-Fender | figures keith4_ needs to continue learning through osmosis.... |
| 04:33.53 | [TK]D-Fender | whamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAMwhamWHAM |
| 04:33.57 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
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| 04:36.41 | Kobaz | [TK]D-Fender: how's that windmill working out for you |
| 04:37.06 | [TK]D-Fender | Quixote... IS my middle name..... right next to DANGEROUS |
| 04:37.20 | keith4_ | rotates slowly in the wind |
| 04:37.56 | keith4_ | if only osmosis worked that way... that'd be nice |
| 04:38.29 | keith4_ | then people could just print out The Book and stuff a pillow with it |
| 04:39.14 | phix | hey I want to do some stuff with detecting different ring tones |
| 04:39.37 | phix | what is the debug command so it shows me the cadence or whateber it is called when an incoming call comes in |
| 04:40.13 | Kobaz | phix: you have a distinctive ring service? |
| 04:40.40 | phix | yeah, I have two phone numbers assigned to the same phone line, if I ring one number it has a different ring than the other |
| 04:41.37 | *** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-839ffc8a5fc337a5) |
| 04:42.08 | phix | Kobaz: this is setup with my PSTN provider |
| 04:42.47 | Kobaz | you may have to monkey with the zap/dahdi code... as far as i know (and i don't know much about the internals) all asterisk knows is that it's 'ringing' |
| 04:43.08 | keith4_ | http://www.voip-info.org/wiki-Asterisk+ZAP+channels |
| 04:43.19 | keith4_ | crap |
| 04:43.23 | keith4_ | rather: http://www.voip-info.org/wiki-Asterisk+ZAP+channels#DistinctiveRingStyles |
| 04:43.44 | keith4_ | sighs |
| 04:43.47 | keith4_ | next topic down. sorry |
| 04:43.50 | keith4_ | opens another beer |
| 04:43.52 | Kobaz | oh |
| 04:43.59 | Kobaz | nifty, i never knew that |
| 04:44.33 | keith4_ | first hit on google for "asterisk distinctive ring" |
| 04:44.35 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 04:45.22 | Kobaz | heh |
| 04:45.29 | Kobaz | anyways |
| 04:45.32 | Kobaz | speaking of rings |
| 04:45.55 | Kobaz | why does it take like 3 rings for a sangoma fxo to hand off the call to asterisk |
| 04:46.23 | russellb | it's waiting so it can detect callerid. |
| 04:46.29 | russellb | and that's asterisk, not the card |
| 04:46.33 | Kobaz | ah |
| 04:46.38 | Kobaz | how would i turn that off |
| 04:46.42 | Kobaz | there's no callerid on the lines |
| 04:46.53 | russellb | edit chan_dahdi.conf (or zapata.conf) and disable callerid |
| 04:46.59 | Kobaz | nifty |
| 04:47.00 | russellb | usecallerid=no i think |
| 04:48.54 | *** join/#asterisk alerer (i=markus@AEP-ONE-FIFTY-NINE.MIT.EDU) |
| 04:49.35 | russellb | analog signaling is overrated, anyway |
| 04:49.41 | Kobaz | heh |
| 04:49.48 | Kobaz | analog signalling is annoying as hell |
| 04:50.00 | russellb | yes. |
| 04:50.08 | [TK]D-Fender | russellb: Who actually gives it good ratings? More like disproportionate expectations if you ask me... |
| 04:50.14 | Kobaz | i had to play for hours to get asterisk to play nice with some fxs's from a nortel |
| 04:50.24 | prg3 | for the privacy option in the Dial() app, can you play with it's behavior? It's very close to what I need... |
| 04:50.25 | russellb | [TK]D-Fender: I don't know, it's just fun to say :-) |
| 04:50.38 | [TK]D-Fender | russellb: Ok, I'll buy that! |
| 04:50.52 | Kobaz | i was trying to get hook flash transfer working, and i played with the delay forever till i found what it liked |
| 04:51.08 | russellb | [TK]D-Fender: I also like to randomly propose deprecating analog signaling, and removing support for it. :-p |
| 04:51.14 | Kobaz | and i had to monkey with the dtmf durations too |
| 04:52.05 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 04:52.06 | [TK]D-Fender | russellb: I pass around petitions to ban di-hydrogen oxide :) |
| 04:52.23 | [TK]D-Fender | russellb: Its dangerous and kills thousands every year! |
| 04:52.31 | ScribbleJ | Honestly, there's probably a lot of code in * that can be removed. All this VOIP and telephny stuff, no one use those parts, do they? |
| 04:52.32 | russellb | I've heard that stuff is used by chemical plants, and gets into our lakes and rivers :-( |
| 04:53.20 | rob0 | www.dhmo.org |
| 04:54.02 | rob0 | DHMO is a major factor in drug abuse, and can be lethal by itself. |
| 04:54.45 | [TK]D-Fender | russellb: its INSIDIOUS.... its also marketed as hydrogen hydroxide. Kids have ready access to it. Its in our parks and schools. IT MUST END! |
| 04:55.34 | maxagaz | do you think it's safer to use asterisk on Debian 5.0 or on Ubuntu 8.04 LTS |
| 04:55.36 | maxagaz | ? |
| 04:55.54 | russellb | maxagaz: up to you. Personally, I'd go with Ubuntu. |
| 04:56.01 | rob0 | Asterisk is probably safer in ANY environment, without DHMO. |
| 04:56.12 | Kobaz | plebian! |
| 04:56.12 | russellb | rob0: yes, that is certain. |
| 04:56.20 | maxagaz | rob0, DHM0 ? |
| 04:56.27 | rob0 | www.dhmo.org |
| 04:57.18 | maxagaz | strange joke |
| 04:58.37 | prg3 | I can't live without the stuff, it's great. |
| 04:58.50 | joobie | hey guys.. anyone know if the polycom 320's can interface to ldap or some external source to pull in speed dial / contacts ? |
| 04:59.04 | joobie | i had a look at the web console.. looks like they can only be programmed in from the handset itself |
| 05:00.12 | russellb | you set that stuff up in the provisioning via ftp or whatever |
| 05:00.24 | [TK]D-Fender | joobie: initial load from provisioning. XHTM for MicroBrowser-based listings, or licensed access to LDAP list. |
| 05:00.36 | *** join/#asterisk talntid (n=eric@66.208.251.170) |
| 05:00.39 | talntid | hi :) |
| 05:01.10 | joobie | so you need an additional license to be able to interface to an ldap server? |
| 05:01.21 | drmessano | jaytee |
| 05:01.49 | [TK]D-Fender | joobie: Directly, yes. |
| 05:02.43 | joobie | what is the indirect free option?:P |
| 05:05.02 | drmessano | Grandstream |
| 05:05.09 | [TK]D-Fender | joobie: MicroBrowser <- |
| 05:06.37 | joobie | ahh |
| 05:06.47 | joobie | "The Polycom model 600, 601, and 650 phones have a feature called Microbrowser .. " |
| 05:06.53 | joobie | that's from http://www.voip-info.org/wiki/view/Polycom+Microbrowser |
| 05:06.59 | joobie | might not be supported on the 320 |
| 05:07.04 | [TK]D-Fender | joobie: it is |
| 05:07.13 | joobie | cool |
| 05:07.19 | [TK]D-Fender | joobie: Stop reading the damn WIKI for everything |
| 05:07.25 | joobie | i remember seeing it in the web interface - just waiting for the phone to reboot atm |
| 05:07.28 | joobie | heh |
| 05:07.30 | [TK]D-Fender | joobie: get your admin guide. |
| 05:07.36 | phix | where is my beer keith4_ ? |
| 05:07.55 | [TK]D-Fender | people configuring their Polycom's via teh Web Browser should be dragged out and SHOT |
| 05:08.01 | [TK]D-Fender | Survivors should be shot AGAIN |
| 05:08.10 | phix | what is dahdi? |
| 05:08.20 | denon | new name of zaptel |
| 05:08.21 | russellb | [TK]D-Fender: I have done it before :-( |
| 05:08.22 | joobie | so has someone written a script to query LDAP and pull it out to XHTML? |
| 05:09.39 | drmessano | O.o |
| 05:10.55 | [TK]D-Fender | joobie: Yes look in http://www.yeseveryonewrotecustomstuffyoucanuseasisnoquestionsaskedevenlessiqrequired.com |
| 05:11.10 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 05:12.04 | drmessano | [TK]D-Fender: Has anyone written a DHTML <> PHP <> Asterisk <> IRC <> Cisco thingamajig? |
| 05:12.07 | drmessano | and free |
| 05:12.45 | drmessano | Bonus if it's written in Django |
| 05:12.52 | keith4_ | phix: it's a smithwick's, and it's quite tasty |
| 05:12.57 | keith4_ | you can enjoy it vicariously through me |
| 05:13.15 | drmessano | Bonus++ if it's written in Django, runs on top of Erlang, and emulated Ruby on Rails |
| 05:13.54 | stabler | does anyone run there asterisk server on gentoo? |
| 05:13.54 | [TK]D-Fender | drmessano: ...can I get that in corn-flower blue? |
| 05:14.08 | [TK]D-Fender | stabler: No. |
| 05:14.15 | drmessano | "Make it look like a sunflower" - Steve Jobs |
| 05:14.19 | stabler | why not |
| 05:14.27 | drmessano | Only morons run Gentoo |
| 05:14.35 | drmessano | *shrug* |
| 05:14.41 | drmessano | Sorry, had to |
| 05:14.43 | stabler | drmessano: why do you say that |
| 05:14.48 | drmessano | DISTRO WARZ |
| 05:14.58 | rob0 | I've moved up to Genthrie |
| 05:15.00 | drmessano | stabler: Listen and learn |
| 05:15.10 | russellb | gentoo is just terrible |
| 05:15.16 | [TK]D-Fender | Genpoo! |
| 05:15.28 | drmessano | (this year + 1) = Year of the Linux Desktop |
| 05:15.35 | drmessano | and |
| 05:15.49 | russellb | but in all seriousness, yes, people run it on gentoo. It's Linux, it will work on there just as well as anything else. Just don't use the ebuilds, as AFAIK, they stay pretty out of date |
| 05:15.52 | [TK]D-Fender | stabler: its a &#^$ing distro. * does not CARE. Satisfy its dependencies and it'll run on just about anything. |
| 05:16.02 | drmessano | ( distro you just mentioned ) sucks balls compared to ( distro I am gonna argue for, just because ) |
| 05:16.31 | [TK]D-Fender | stabler: are you skilled at Gentoo? |
| 05:16.41 | stabler | yesh |
| 05:16.43 | drmessano | Ubuntu sucks, real men run Knoppix |
| 05:17.02 | drmessano | Gentoo sucks, real men run Linspire |
| 05:17.08 | stabler | lmao |
| 05:17.17 | drmessano | Debian sucks, real men run Fedora |
| 05:17.28 | drmessano | ^^^^ Its about like that |
| 05:17.37 | russellb | live cds suck, real men run their distro off of a tape drive |
| 05:17.42 | stabler | i just was wondering if there were any weird known issues before i dove in |
| 05:17.52 | russellb | stabler: just that you shouldn't use the ebuilds. |
| 05:18.09 | stabler | russellb: fo sho |
| 05:18.12 | drmessano | russellb: Loser.. I have a distro + asterisk running on a USB floppy drive connected to a WRT54G |
| 05:18.16 | stabler | i saw how old they were =/ |
| 05:18.39 | drmessano | USB 5 1/4, mind you |
| 05:18.50 | drmessano | Hang on, call coming in.. Need to flip the disk |
| 05:18.57 | [TK]D-Fender | drmessano: You had me convinced.... right up till you mentioned 5.25 |
| 05:19.00 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 05:19.02 | drmessano | lol |
| 05:19.19 | drmessano | IVR requires inserting disk 4 of 11 |
| 05:19.23 | [TK]D-Fender | drmessano: And SS DD as well? Teh shiznit yo! |
| 05:19.23 | drmessano | :( |
| 05:19.48 | drmessano | Oh, funnyman wants voicemail.. where the heck is disk 8? |
| 05:20.10 | drmessano | I need a floppy RAID |
| 05:20.29 | stabler | floppy raid.. thats great |
| 05:20.33 | *** join/#asterisk eharris (i=eharris@99-179-7-82.lightspeed.austtx.sbcglobal.net) |
| 05:20.53 | drmessano | I need to find that page where some guy had 5 3.5" USB floppy drives in a RAID5.. |
| 05:21.01 | drmessano | 4MB array |
| 05:21.03 | stabler | drmessano: in all seriousness i think you should try to get a functioning floppy raid going |
| 05:21.52 | drmessano | http://ohlssonvox.8k.com/fdd_raid.htm |
| 05:22.34 | [TK]D-Fender | OMMFG |
| 05:23.40 | stabler | i love it! |
| 05:24.15 | drmessano | Wanna piss that dude off? |
| 05:24.19 | drmessano | Go to his house |
| 05:24.25 | drmessano | Hit the eject buttons |
| 05:24.27 | drmessano | and laugh |
| 05:24.31 | [TK]D-Fender | drmessano: With a bulk tape eraser :) |
| 05:24.33 | drmessano | "REBUILD THAT!!!" |
| 05:24.48 | *** join/#asterisk Gopaul (n=chatzill@61.95.156.130) |
| 05:25.10 | drmessano | -5 because he used a MAC, not a real computer |
| 05:25.15 | drmessano | But otherwise, cool |
| 05:25.27 | *** join/#asterisk CunningPike (n=arodgers@S01060014bf81366b.vc.shawcable.net) |
| 05:25.58 | russellb | "I plugged stuff in and hit the RAID button." |
| 05:25.59 | joobie | [TK]D-Fender, thanks for the url |
| 05:26.09 | russellb | yeah, a real feat |
| 05:26.17 | russellb | funny, anyway, though |
| 05:26.45 | drmessano | "Where did the rest of the space go? Dunno, RAID??" |
| 05:26.53 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 05:26.53 | *** mode/#asterisk [+o denon] by ChanServ |
| 05:27.15 | drmessano | -5 more for not knowing how a RAID works, yet he created a RAID intentionally |
| 05:27.34 | drmessano | "I accidentally the whole DD-WRT on my router" |
| 05:27.42 | drmessano | Yep, I do that all teh time!!!! |
| 05:30.30 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 05:30.40 | ricko73 | RAID's like a backup system right? |
| 05:30.46 | ricko73 | ducks |
| 05:31.06 | drmessano | Yes |
| 05:31.08 | drmessano | Now |
| 05:31.12 | prg3 | ricko73: sure it is.. when the raid's on another machine, and has multiple copies for historical.. |
| 05:31.20 | drmessano | Go to the RAID thingo, and take one of the drives home for backup |
| 05:31.26 | drmessano | Pick one, any one |
| 05:31.39 | ricko73 | except the hot spare |
| 05:31.40 | ricko73 | ;) |
| 05:31.43 | drmessano | prg3: He was making a joke |
| 05:31.50 | drmessano | hot spare? |
| 05:31.55 | joobie | damnit |
| 05:31.55 | drmessano | I dunno wut that r |
| 05:32.03 | joobie | cant find a script to convert ldap microsoft to XHTML |
| 05:32.04 | prg3 | drmessano: as was I.. I guess i shoulda delivered better.. |
| 05:32.16 | drmessano | ROFL |
| 05:32.26 | joobie | luckily ive been playing with .net c# lately.. |
| 05:32.33 | prg3 | And it wasn't funny.. |
| 05:32.44 | drmessano | You're gonna enumerate LDAP in AD and pull that into XHTML? |
| 05:32.45 | ricko73 | heads to get another beer |
| 05:32.47 | drmessano | Good luck with that |
| 05:33.01 | drmessano | Sounds like 2 or 3 lines a batch file, or something |
| 05:33.43 | drmessano | http://madetoordercustomscriptsforfreelibrarywiki.com |
| 05:34.39 | drmessano | Expect to have to write code here |
| 05:34.48 | *** join/#asterisk Maliuta (n=scooby@kiev.lusan.id.au) |
| 05:36.25 | *** join/#asterisk voxter (n=voxter@S0106001c1025ca09.vc.shawcable.net) |
| 05:45.34 | [TK]D-Fender | joobie: its XHTML. its basic friggen tags. jsut code it. |
| 05:52.29 | the_5th_wheel | Has anyone tried running two seperate versions of asterisk on one box? |
| 05:54.11 | the_5th_wheel | Cause I just cant get asterisk 1.4(Bristuff) stabally. And I need 1.4 for my custom agi scripts |
| 05:54.24 | *** join/#asterisk Mw3 (n=mw3@ip59934bd1.rubicom.hu) |
| 05:55.18 | joobie | [TK]D-Fender, ya.. that's the easy part |
| 05:55.25 | joobie | harder part is pulling it from AD |
| 06:02.51 | *** join/#asterisk harry_v (n=pcsuppor@S0106001d7e52cc78.vs.shawcable.net) |
| 06:04.02 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 06:04.04 | [TK]D-Fender | checkout time, later all |
| 06:10.11 | ghento | Hi all. I'm calling out to numbers using SIP. Within my extensions.conf I execute a Read() asking for one input, however when I press a number, it is never recognized. I'm viewing the call executions in the console with verbosity set high, and no user input seems to be accepted. Any thoughts on what could be causing this? |
| 06:22.47 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 06:56.14 | *** join/#asterisk killown (n=Yamato@unaffiliated/killown) |
| 06:57.18 | *** join/#asterisk scientes (n=scientes@75-165-95-28.tukw.qwest.net) |
| 07:01.26 | *** join/#asterisk brunner (n=chris@68-119-87-106.dhcp.mtgm.al.charter.com) |
| 07:01.45 | brunner | what much do I get charged when I have a toll free number when people call from a pay phone? |
| 07:01.53 | brunner | s/what/how |
| 07:02.06 | MaliutaLap | brunner speak to your carrier |
| 07:02.46 | brunner | MaliutaLap: isn't there a uniform payphone surcharge in the US? |
| 07:03.06 | MaliutaLap | well that assumes everyone here is _in_ the us |
| 07:03.14 | MaliutaLap | and not all the countries of the world |
| 07:03.34 | MaliutaLap | and all PSTN charges would be determined by the carrier |
| 07:03.43 | MaliutaLap | as would all VoIP charges |
| 07:04.37 | MaliutaLap | you can use SIP through my server ... I'll charge you $100US/min charged to the nearest 5min block |
| 07:05.55 | drmessano | brunner: Tollfree charges are per the CARRIER not the calling partys rate |
| 07:06.15 | brunner | yeah, got it. thanks. |
| 07:11.54 | *** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun) |
| 07:14.44 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 07:16.13 | MaliutaLap | yes! jbot can play with me! ;) |
| 07:25.01 | *** join/#asterisk grEvenX (n=even@apb9hb.ip.ssc.net) |
| 07:26.35 | *** join/#asterisk _gm (n=gmustafa@115.186.106.37) |
| 07:28.26 | brunner | what's the correct syntax for the Async parameter of the Originate command? I can't find it on voip-info, in the book, or on google |
| 07:28.45 | brunner | do I just add "Async" on it's own line, or "Async: yes" or "Async: true" or what? |
| 07:29.06 | brunner | maybe "Async: on"? |
| 07:34.19 | *** join/#asterisk botox93 (n=botox93@213.221.82.242) |
| 07:34.37 | *** join/#asterisk BeeBuu (n=beebuu@116.20.180.99) |
| 07:35.00 | *** join/#asterisk xrmx__ (n=rm@host221-177-dynamic.25-79-r.retail.telecomitalia.it) |
| 07:35.04 | BeeBuu | 120 channels need which kind CPU? |
| 07:35.13 | brunner | well, I'm obviously doing it wrong, because I'm not getting an OriginateResponse event |
| 07:35.13 | BeeBuu | ISDN |
| 07:37.13 | brunner | BeeBuu: I prefer AMD64 or i386, but there are probably ten other architectures you could compile asterisk on. The choice is yours. |
| 07:38.45 | brunner | BeeBuu: I have friends who have lots of nice things to say about Sun SPARC |
| 07:39.14 | BeeBuu | em... |
| 07:40.10 | MaliutaLap | sun runs on x86 kit these days |
| 07:40.29 | brunner | hmm, cool |
| 07:43.43 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:52.41 | ScribbleJ | So does ASterisk compile on non x86 platforms? |
| 07:53.38 | ScribbleJ | I wrote a plugin to * that does some net comms, I didn't do anything about byte order, guess I'll have to go back and do that if * is also run on non-x86. |
| 08:07.47 | *** join/#asterisk tamiel (n=tamiel@213.30.183.226) |
| 08:12.04 | *** join/#asterisk frk2 (n=frk2@zivios/member/fkhan) |
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| 08:26.28 | *** join/#asterisk DragoraN (n=DragoraN@dial-248-231.orange.sk) |
| 08:26.33 | Damin | oej: morning.. |
| 08:27.05 | *** join/#asterisk iamy_china (n=iamy_chi@123.117.178.115) |
| 08:27.39 | iamy_china | Hoho |
| 08:27.56 | iamy_china | Finally got in |
| 08:29.26 | *** part/#asterisk iamy_china (n=iamy_chi@123.117.178.115) |
| 08:30.03 | *** join/#asterisk iamy_china (n=iamy_chi@123.117.178.115) |
| 08:32.56 | iamy_china | hello? |
| 08:33.20 | MaliutaLap | goodbye |
| 08:33.33 | iamy_china | cool |
| 08:45.30 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 08:49.20 | *** join/#asterisk gr0mit (n=tim@CSC.ge4-0-0.401.ar1.BBS1.gblx.net) |
| 08:58.37 | oej | Damin: Morning. Up early? |
| 09:04.23 | *** join/#asterisk medjr (n=medjr@41.226.181.96) |
| 09:06.03 | *** join/#asterisk LuisTorres (n=chatzill@a213-22-94-93.cpe.netcabo.pt) |
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| 09:22.16 | iamy_china | It's afternoon here |
| 09:33.17 | *** join/#asterisk jermey_g (n=me@static-213-115-44-229.sme.bredbandsbolaget.se) |
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| 09:36.57 | *** join/#asterisk iamy_china (n=iamy_chi@123.117.178.115) |
| 09:41.34 | norgs | Hi, I have a quick question. I'm having some problem with parked calls |
| 09:44.01 | norgs | I'm trying to use FOP to park a call by dragging and dropping it's icon onto the parking slot, but it returns an error to me: |
| 09:44.10 | norgs | Starting SIP/108-096708a0 at parkedcalls,701,1 failed so falling back to exten 's' |
| 09:45.13 | norgs | Am I supposed to supply a parkedcalls context? |
| 09:49.49 | jermey_g | norgs:show us your config |
| 09:49.51 | *** join/#asterisk sergee (n=serg@voip1.west-call.com) |
| 09:50.21 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
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| 09:51.25 | *** join/#asterisk mahiti-irc (n=mahiti1@121.243.168.201) |
| 09:51.33 | mahiti-irc | hi all |
| 09:51.48 | mahiti-irc | need help on setup IVR |
| 09:52.09 | *** join/#asterisk rdlang (n=dlangr@ns03.ism.nl) |
| 09:52.22 | mahiti-irc | i have configured server with AsteriskNow and TDM400P card |
| 09:52.55 | norgs | jermey_g: dear lord, you don't want that it's half elastixed, half freepbxed, half upgraded to asterisk 1.6 and three quarters hacked up |
| 09:53.00 | mahiti-irc | can anyone help on this ? |
| 09:54.35 | rdlang | hi, i was wondering, i set up agents.conf, queue.conf and entered a line in my extensions.conf "exten => 8891,1,AgentLogin(1001)" , but when i call 8891 it say's it does not have an application AgentLogin for that extension |
| 09:55.06 | rdlang | i also loaded app_queue.so |
| 09:55.10 | mvanbaak | norgs: maybe this problem is a good motivation to cleanup the mess ? |
| 09:55.33 | norgs | i'm motivated enough, time is the problem |
| 09:56.02 | brunner | what's the proper syntax for passing more than one argument with AGI? AGI(command|args) <== |
| 09:56.13 | norgs | i need to get people out of my hair so i can spend some time cleaning, but i won't get that until i get it working |
| 09:56.48 | *** join/#asterisk Mr_BOnD_007 (i=Mr_BOnD_@119.160.199.6) |
| 09:58.45 | rdlang | does someone know why i get this: pbx_extension_helper: No application 'AgentLogin' for extension (default, 8891, 1) |
| 09:59.11 | brunner | can I pass ${THISNUM}&${UID}? |
| 10:00.38 | *** join/#asterisk DragoraN (n=DragoraN@95.105.137.64) |
| 10:01.58 | *** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog) |
| 10:07.23 | *** join/#asterisk scr (i=lubo@zatwor.sk) |
| 10:08.17 | Gopaul | when we use originate method for a call how we know the call is answered or not? |
| 10:15.45 | hi365 | argh!!! whats up with iax2 and status UNREACHABLE? |
| 10:18.18 | *** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net) |
| 10:22.54 | mvanbaak | hi365: what asterisk versions on both ends ? |
| 10:23.39 | hi365 | mvanbaak: 1.4.22 <=> 1.2.22 |
| 10:24.06 | hi365 | or 1.4.22 <=> teliax |
| 10:24.32 | hi365 | (heck on the 1.4.22, even the connections to iaxmodem are unknowen) |
| 10:27.05 | *** join/#asterisk dumezil (n=who@219.142.224.119) |
| 10:27.28 | mvanbaak | hhmm |
| 10:27.53 | mvanbaak | I know there are some problems with it between specific versions of 1.4 talking to specific versions of 1.6 |
| 10:27.58 | mvanbaak | but that's not the case here |
| 10:28.12 | hi365 | reload, reboot, smapck on the head, kick in the pants, still no go :( |
| 10:28.27 | hi365 | it worked till a few weeks ago - cant think of anything that changed |
| 10:29.35 | dumezil | im trying to setup a tdm400p with asterisk 1.4.23. the driver seems to work (dahdi_hardware gives me "pci:0000:00:09.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F"), but i can't seem to get it to work with asterisk. within asterisk "dahdi show status" gives me "Wildcard TDM400P REV E/F Board 5", but dahdi show channels doesn't show anything. does anyone know how to correctly configure dahdi? |
| 10:31.07 | *** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com) |
| 10:31.07 | stmaher | hello everyone.. |
| 10:31.15 | stmaher | Im having trouble recording to a file.. |
| 10:31.38 | stmaher | exten => _200,3,Record(/tmp/newgreeting:wav) |
| 10:31.49 | stmaher | Executing [200@int-hardphones:3] Record("SIP/7301-08c4d988", "/tmp/newgreeting:wav") in new stack |
| 10:31.56 | stmaher | but no file is created in /tmp/ at all . |
| 10:32.07 | stmaher | I have tried all codes ulaw wav sln etc.. but none work. |
| 10:32.18 | stmaher | no error is produced.. and /tmp has all the right permissions.. |
| 10:32.21 | stmaher | am i missing something? |
| 10:32.23 | *** join/#asterisk lbt (n=david@78.32.229.233) |
| 10:37.23 | tzafrir_laptop | dumezil, what's the output of lsdahdi #? |
| 10:44.44 | dumezil | ### Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) |
| 10:44.52 | dumezil | <PROTECTED> |
| 10:44.56 | dumezil | <PROTECTED> |
| 10:45.01 | dumezil | <PROTECTED> |
| 10:45.07 | dumezil | <PROTECTED> |
| 10:45.36 | dumezil | ports 1 and 2 are fxs and 3 and 4 are fxo |
| 10:50.39 | *** join/#asterisk gulden (n=gulden@89.214.149.220) |
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| 10:59.09 | hi365 | anyone have a link on how iax2 oppertates, so that I can try to understnd the (alomost bare) debug? |
| 11:00.34 | *** join/#asterisk jksM (i=jks@193.189.93.254) |
| 11:01.42 | hi365 | chan_iax2.c: Received error: Bad file descriptor <--------- hu? |
| 11:02.48 | tzafrir_laptop | dumezil, it is configured, but asterisk doesn't use it |
| 11:02.59 | tzafrir_laptop | Next you should check zapata.conf / chan_dahdi.conf |
| 11:03.25 | tzafrir_laptop | What is the output of: asterisk -rx 'dahdi show channels' |
| 11:04.29 | tzafrir_laptop | hi365, IIRC iax2 basically uses a single socket |
| 11:04.40 | hi365 | nu? |
| 11:04.45 | tzafrir_laptop | it was unable to bind to the port or something similar? |
| 11:04.54 | hi365 | so that means the socket is busy? |
| 11:05.15 | hi365 | ah |
| 11:05.15 | hi365 | wonders why would that be |
| 11:06.52 | *** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net) |
| 11:06.58 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 11:12.13 | hi365 | tzafrir_laptop: seems aterisk isnt listening on 4569? |
| 11:17.27 | *** part/#asterisk gulden (n=gulden@89.214.149.220) |
| 11:21.57 | jermey_g | is running IP protocol over a PRI is a waste? |
| 11:22.06 | jermey_g | a waste |
| 11:23.08 | jermey_g | do you have to run a similar protocol such as PPP is ran over PSTN lines and on top is IP to get a connection up e.g. as in dial-up (mgetty,ppp) |
| 11:37.49 | *** join/#asterisk jmacz (n=jmacz@190.25.41.170) |
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| 11:46.49 | *** join/#asterisk error404notfound (n=error404@58-65-160-128.nayatel.pk) |
| 11:47.24 | error404notfound | any comments on "[Mar 3 16:45:06] NOTICE[1034] chan_iax2.c: Auto-congesting call due to slow response" in messages log and not able to dial any number outside that subnet? |
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| 12:16.31 | Milad | Hi , I add some localization for fa_ir in asterisk, can you guide me how to public it ? |
| 12:18.47 | *** join/#asterisk msetim (n=setim@200.195.162.210) |
| 12:19.38 | msetim | There are someone using asterisk 1.4.23.1 with updatecdr='yes' in agents.conf? |
| 12:21.02 | *** join/#asterisk erth64net (n=erth64ne@96-25-65-141.war.clearwire-wmx.net) |
| 12:25.27 | *** join/#asterisk uluatu (n=isolve@200.195.162.210) |
| 12:27.48 | uluatu | why 1.4.23.1 is not updating the cdr userfield with the recording name of an agent? I already set the updatecdr parameter in agents.conf |
| 12:34.24 | jermey_g | uluatu:pastebin your configs |
| 12:36.32 | uluatu | jermey_g: hi, we have a bug fotr that on http://bugs.digium.com/view.php?id=14590 |
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| 12:43.06 | the_5th_wheel | Has anyone tried running two seperate versions of asterisk on one box? Cause I need asterisk 1.4 for my agi, but the bristuff for 1.4 bombs on me twice a day... |
| 12:43.41 | uluatu | the strange thing is that when calling the agent directly the cdr is fine, the recording filename is there. When the call comes from a queue, no CDR for userfiled |
| 12:48.55 | uluatu | jermey_g: are you using that feature? |
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| 12:53.28 | jermey_g | uluatu:not at present. |
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| 12:59.17 | *** join/#asterisk xezz (n=asdasd@serial.hq.ondsl.gr) |
| 13:00.51 | xezz | hello, is it possible for asterisk to callback and play a set of recorded questions while user answers ? |
| 13:01.46 | xezz | like "Please say your name" user answers then "Please say your surname" etc... |
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| 13:11.41 | jermey_g | xezz:both are possible |
| 13:11.49 | xezz | k cool |
| 13:12.00 | xezz | can you suggest a way ? |
| 13:12.31 | jermey_g | xezz:use Background to playback your questions |
| 13:12.47 | jermey_g | while use any of the asterisk speech recognition engine |
| 13:12.55 | jermey_g | festival or sth |
| 13:13.29 | xezz | hmm |
| 13:13.50 | xezz | is there a web gui like freepbx or anything else i could use to begin with testing ? |
| 13:13.57 | xezz | or it should be via cli |
| 13:14.41 | jermey_g | via cli |
| 13:15.01 | jermey_g | there may be a gui but it ll be commercial |
| 13:15.20 | xezz | speech recognizers are language independent ? |
| 13:15.36 | jermey_g | they are language dependent offcourse |
| 13:16.35 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 13:17.04 | xezz | hmm then i think i will have problem with Greek language :/ |
| 13:17.30 | mort_gib | xezz: Like most people do :-) |
| 13:18.28 | xezz | heh |
| 13:18.41 | xezz | nah, english is worldwide |
| 13:18.44 | jermey_g | i believe that SRs are available for popular langs |
| 13:19.27 | jermey_g | unless the lang is spoken in 3rd world or only known to UN lang extinction dept |
| 13:20.17 | xezz | yes |
| 13:20.26 | xezz | if i use callback function |
| 13:20.54 | xezz | can asterisk callback the user and then playback a recorded question while it will record user's answers ? |
| 13:21.02 | mort_gib | xezz: I'm actually Danish, so... |
| 13:21.24 | xezz | ah, welcome to the club of unsupported then bro :) |
| 13:22.49 | mort_gib | xezz: :-) asterisk can do callback, not a problem, but you have to go beyond extensions.conf agi or similar |
| 13:24.05 | [TK]D-Fender | mort_gib: huh? |
| 13:24.29 | [TK]D-Fender | xezz: record what kind of answer? |
| 13:26.08 | mort_gib | [TK]D-Fender: If you want asterisk to callback based on an incoming call, and you want to control the message played back you would need some additional functionality... I know how much you like extensions.conf but surely you would need more ;-) |
| 13:26.22 | xezz | oh basically it doesnt have to have greek language support, english is fine :) |
| 13:26.37 | [TK]D-Fender | mort_gib: No, you wouldn't |
| 13:27.01 | xezz | [TK]D-Fender, im looking to integrate voice recognition |
| 13:27.22 | mort_gib | Uhm, maybe I misunderstood xezz... |
| 13:27.33 | xezz | the scenario is: asterisk call back user, plays a recorded question, user answers, asterisk asks again, user answers, etc... |
| 13:27.50 | xezz | maybe my english is so badmort ^^ |
| 13:28.22 | [TK]D-Fender | xezz: For calculated responses you would use AGI. Simple ones along a linear chain could be done in straight dialplan. |
| 13:28.50 | mort_gib | xezz: No not at all, maybe spaghetti coding in PHP while typing in here is a no go |
| 13:28.58 | [TK]D-Fender | xezz: There are a few VR packages out there that can be used with *. |
| 13:33.57 | mort_gib | [TK]D-Fender: so agi it is :-) |
| 13:34.10 | [TK]D-Fender | mort_gib: Not yet... |
| 13:34.30 | mort_gib | [TK]D-Fender: Stubborn today :-) |
| 13:34.50 | mort_gib | [TK]D-Fender: too early for you! |
| 13:34.51 | [TK]D-Fender | mort_gib: I object.... what do you mean "today"? :p |
| 13:35.22 | mort_gib | [TK]D-Fender: true |
| 13:36.01 | xezz | hmmm thank anyway |
| 13:36.04 | xezz | thanks* |
| 13:36.28 | [TK]D-Fender | xezz: There are a few VR apps listed on the WIKI include CMU Sphinx which is free |
| 13:36.49 | [TK]D-Fender | xezz: There are a few others with fairly affordable by channel pricing |
| 13:39.24 | xezz | yes thank you |
| 13:39.40 | xezz | im googling for something that might have freepbx support also |
| 13:40.05 | xezz | but no luck till now :| |
| 13:43.09 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 13:43.26 | [TK]D-Fender | xezz: What does "FreePBX support" mean? |
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| 13:43.43 | [TK]D-Fender | xezz: this is custom stuff no matter what. Nothing in FreePBX should interfere |
| 13:44.08 | xezz | yep i meant something with gui |
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| 13:46.21 | mark17 | hello, what are good voip/sip providers with free calls to numbers in NL and other parts of the world? |
| 13:49.08 | [TK]D-Fender | xezz: three are completely separate solutions for this. GOOD LUCK |
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| 13:54.31 | xezz | thanks |
| 13:57.33 | [TK]D-Fender | useless GUI twits..... FTW! |
| 13:58.13 | *** join/#asterisk Dr-Linux|home (n=Nothing@119.63.130.34) |
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| 14:04.22 | tzafrir_laptop | Have an extra computer to kill? Try Bricker^WBriker - http://briker.org/ |
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| 14:05.01 | tzafrir_laptop | seems to be based in Ubuntu, with freepbx and a2billing installed |
| 14:05.46 | Dr-Linux|home | i want to provide a menu while caller is waiting in the queue? but once agents gets free caller show direct to that agent. Is it possible? |
| 14:06.14 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:06.55 | [TK]D-Fender | Dr-Linux|home: Go read the sample queues.conf |
| 14:07.20 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 14:07.29 | Dr-Linux|home | [TK]D-Fender: Answer is not available there for my question |
| 14:08.12 | [TK]D-Fender | Dr-Linux|home: You want them to wait for agents. that is a queue. in a queue you can present them a menu from which they can quit the queue to do something else instead. |
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| 14:10.14 | Dr-Linux|home | [TK]D-Fender: yes, they will have options while waiting like "press this for news headlines" he listening to news or songs but as agent is available song/news is quit and he is routed to agent |
| 14:10.29 | [TK]D-Fender | Dr-Linux|home: if you want them to navigate a whole pile of IVR, etc, then before letting them move onto that, originate a call that goes into the queue holding the chann name of who to hijack, and whent he agent answers it they have the option to grab the caller |
| 14:10.29 | mcnobody | Hello everyone. I have little AEL syntax error problem. http://pastebin.com/d46766286 |
| 14:11.32 | russellb | mcnobody: try putting quotes around the value in the assignment |
| 14:13.48 | codefreeze-lap | mcnobody: I think I may have fixed that real recently. Try the svn co of 1.6.0, and see if it's still a problem. Hmmm, I'll try it here, and see also. |
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| 14:14.44 | mcnobody | codefreeze-lap: ok. |
| 14:15.17 | Dr-Linux|home | [TK]D-Fender: thanks - i was thinking if asterisk gives any feature for that |
| 14:15.31 | mcnobody | russellb: That actually works, but the number variable has then also quote characters in. |
| 14:15.56 | russellb | ok, well codefreeze-lap is the AEL master, so we'll see what he says :-) |
| 14:16.25 | mcnobody | Okey. |
| 14:16.40 | *** part/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 14:17.39 | [TK]D-Fender | Dr-Linux|home: and I jsut gave you the answer |
| 14:18.36 | mcnobody | codefreeze-lap: The first AEL settings generates Set(number=$[${GLOBAL(ULKOPREFIX)}9${x}]) to dialplan which works I have excepted. But there is the warning with Asterisk 1.6.0.6. |
| 14:19.11 | [TK]D-Fender | umm... how is that an expression? |
| 14:19.56 | mcnobody | I have not time to test it today with svn version myself. Perhaps tomorror. |
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| 14:20.31 | Kobaz | mcnobody: yeah, what the heck is that? you should fix it |
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| 14:21.03 | Dr-Linux|home | [TK]D-Fender: my 2nd question is, when caller is listening to moh and as call is assigned to an agent and that agents is rining .. now caller should stop listening to moh and listen to ring? how it will possible? |
| 14:21.40 | mcnobody | Kobaz: It's generated from AEL code: number=${GLOBAL(ULKOPREFIX)}9${x}; See http://pastebin.com/d46766286 |
| 14:21.43 | [TK]D-Fender | Dr-Linux|home: Read the sample config <- |
| 14:22.07 | Dr-Linux|home | [TK]D-Fender: I read but couldn't find for this |
| 14:22.43 | yang | Could someone point me to a good solution for having QOS-ed network based on linux for preferencing VOIP packets |
| 14:22.59 | Kobaz | mcnobody: [Mar 3 15:57:16] WARNING[14085]: ast_expr2.fl:435 ast_yyerror: ast_yyerror(): file /etc/asterisk/extensions.ael, line 374, columns 13-48, variable |
| 14:23.02 | Kobaz | mcnobody: do you not see that? |
| 14:23.30 | [TK]D-Fender | Dr-Linux|home: Keep reading |
| 14:23.49 | Dr-Linux|home | :) |
| 14:23.53 | codefreeze-lap | mcnobody: OK, I see the same things even with the latest. Sigh. Need to update ast_expr2.fl again. In the meantime, remember these are just warnings. Won't affect your output at all. |
| 14:23.58 | Kobaz | mcnobody: use quotes |
| 14:23.58 | mcnobody | Kobaz: Yes, Ican see. |
| 14:24.17 | Kobaz | mcnobody: that's an invalid line, which is why you have screwy dialplan code being made |
| 14:24.25 | mcnobody | codefreeze-lap: Thanks. |
| 14:24.57 | codefreeze-lap | Kobaz: mcnobody: you MIGHT be able to get around this with quoting, but... I think/hope I can find a better way... |
| 14:26.12 | mcnobody | codefreeze-lap: The quoting helps but then variable number has quote character at the begining and at the end of the value. |
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| 14:27.05 | mcnobody | corefreeze-lap: I know it works like I have excepted but I just wanted to know if the syntax is wrong and how I should write it the correct way. |
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| 14:28.19 | mcnobody | Kobaz: Could you help to get that AEL line as valid AEL line? |
| 14:28.23 | codefreeze-lap | mcnobody: yeah, I think I can tweak the grammar and get to accept things as they are. The goal of the check being performed is to spot problems in your expressions before you feed them into asterisk (and have runtime probs). |
| 14:29.31 | Kobaz | mcnobody: quote it |
| 14:29.32 | mcnobody | codefreeze-lap: Thanks. I understood. Do you think thats valid AEL line? |
| 14:29.52 | codefreeze-lap | mcnobody: Yes, it should be. |
| 14:30.31 | Dr-Linux|home | [TK]D-Fender: i read again but couldn't find the solution for my question :) |
| 14:30.31 | mcnobody | Kobaz: Does every AEL variable setting need quotes? |
| 14:30.36 | Kobaz | number="${GLOBAL(ULKOPREFIX)}9${x}"; number="512059${x};" |
| 14:30.52 | Kobaz | mcnobody: no, just ones with variable concatentation that i've noticed |
| 14:31.19 | [TK]D-Fender | mcnobody: I don't see you calling SET <- |
| 14:31.31 | [TK]D-Fender | mcnobody: That isn't a GLOBAL you're doing there |
| 14:31.40 | [TK]D-Fender | mcnobody: from what I see you still need the APP |
| 14:32.18 | codefreeze-lap | mcnobody: it's not a problem at runtime because all the ${...} expressions are evaluated out of the expression before its passed to the ast_expr stuff. |
| 14:32.41 | [TK]D-Fender | codefreeze-lap: Sample config still shows use of Set as an app, not free-form "X=y" |
| 14:33.04 | codefreeze-lap | and, as to quotes, the MAIN reason to use them is when spaces might be involved, to keep things to a single 'token' |
| 14:33.04 | Kobaz | [TK]D-Fender: x=y is valid though |
| 14:33.18 | [TK]D-Fender | Kobaz: Where would I see that documented? |
| 14:33.27 | Kobaz | [TK]D-Fender: it's in there somewhere |
| 14:33.48 | codefreeze-lap | [TK]D-Fender: the voip-info wiki documents it. |
| 14:34.01 | [TK]D-Fender | Kobaz: I'm reading now... I don't see it yet. And the WIKI is untrustworthy |
| 14:34.14 | Kobaz | [TK]D-Fender: http://www.voip-info.org/wiki/view/Asterisk+AEL2 |
| 14:34.16 | codefreeze-lap | [TK]D-Fender: Uh, at least, it had *better*! ;) |
| 14:34.32 | Kobaz | [TK]D-Fender: scroll down to context foo { 555 => |
| 14:34.40 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 14:35.01 | Kobaz | NOTE: AEL wraps the right hand side of an assignment with $[ ] to allow expressions to be used If this is unwanted, you can protect the right hand side from being wrapped by using the Set() application. Read the README.variables about the requirements and behavior of $[ ] expressions. |
| 14:35.24 | Kobaz | mcnobody: what you really want, is to use Set in the ael |
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| 14:35.47 | riddlebox | Katty, you frozen? |
| 14:35.49 | mcnobody | [TK]D-Fender: AEL2 BNF at http://www.voip-info.org/wiki/view/Asterisk+AEL2 really says that x=1 is valid. |
| 14:35.55 | Kobaz | licks KavanS |
| 14:36.01 | Kobaz | licks Katty |
| 14:36.11 | Kobaz | ...yeap, ice cold |
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| 14:36.23 | [TK]D-Fender | mcnobody: Go show use what is parsed in "dialplan show" |
| 14:36.36 | Kobaz | [TK]D-Fender: it converts it to a Set($[... |
| 14:36.37 | codefreeze-lap | Kobaz: mcnobody: yes, he could wrap both statements in Set(...) and get rid of the complaints, that is true. |
| 14:36.57 | [TK]D-Fender | Kobaz>[TK]D-Fender: it converts it to a Set($[... <-- BROKEN |
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| 14:37.53 | Kobaz | [TK]D-Fender: er i mean... its Set(foo=$[... |
| 14:38.21 | [TK]D-Fender | McShow us what gets parsed |
| 14:38.29 | ManxPowerAsteria | Yes, X=1; is valid in AEL |
| 14:38.48 | vlt | Hello. Can anyone reccomend a small asterisk distribution that supports HFC-S hardware and fits on a 512 MB flash hard drive? |
| 14:39.25 | mcnobody | [TK]D-Fender: AEL number=${GLOBAL(ULKOPREFIX)}9${x}; is parsed to Set(number=$[${GLOBAL(ULKOPREFIX)}9${x}]) |
| 14:39.58 | [TK]D-Fender | mcnobody: PASTEEBIN |
| 14:40.05 | Kobaz | that's all you need |
| 14:40.20 | Kobaz | convert your ael to be: Set(number=${GLOBAL(ULKOPREFIX)}9${x}); |
| 14:41.52 | codefreeze-lap | mcnobody: Kobaz: yep, and in the meantime, I'll see if I can make ast_expr2 not complain. |
| 14:43.07 | Kobaz | codefreeze-lap: what is the behavior of $[] if one of the variables expands to a string |
| 14:43.21 | ManxPowerAsteria | $[] says "This is math" |
| 14:43.24 | Kobaz | i think i've used it once |
| 14:43.30 | Kobaz | er |
| 14:43.35 | Kobaz | wrong chan... but anyway |
| 14:43.39 | Kobaz | yeah i know that's for math |
| 14:43.43 | Kobaz | but what the behavior |
| 14:44.50 | Kobaz | is the whole thing set to 0 or what... |
| 14:44.54 | Kobaz | i should try it out to see |
| 14:45.30 | ManxPowerAsteria | Watching the CLI should tell you. |
| 14:47.16 | [TK]D-Fender | ManxPowerAsteria: I asked to see the "dialplan" show but people here see to have issues following simple debugging requests. |
| 14:47.19 | mort_gib | any recommendations for a "reception phone" ?? |
| 14:47.41 | Kobaz | mort_gib: polycom 650 with sidecar |
| 14:48.00 | mort_gib | Kobaz: Yeah! |
| 14:48.05 | [TK]D-Fender | mort_gib: Managing how many other phones? |
| 14:48.48 | codefreeze-lap | Kobaz: It should return the string... The only exception I can think of, it that might play havoc on numeric strings... like 000002 might turn into 2. That sort of thing. In those kinds of cases, use Set(). |
| 14:48.58 | mort_gib | biggest install so far is some 40 phones, where about 20 should "appear" on the reception phone |
| 14:50.39 | codefreeze-lap | Kobaz: and usually, the use of some operator would be required to get that to happen. |
| 14:50.44 | Kobaz | codefreeze-lap: x=y should probably either throw a warning when used with a non-expression, or it should dump the $[] if it detects a non-expression... i vote for the ladder |
| 14:51.03 | Kobaz | latter |
| 14:51.33 | [TK]D-Fender | slams Kobaz with a ladder and tags ManxPowerAsteria into the ring |
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| 14:51.54 | [TK]D-Fender | mort_gib: IP 650 + 2 sidecars |
| 14:51.55 | Kobaz | stupid english |
| 14:51.58 | ManxPowerAsteria | [TK]D-Fender: if they didn't have trouble following instructions they would not be here asking questions. |
| 14:52.08 | Kobaz | mort_gib: THREE sidecars! |
| 14:52.11 | codefreeze-lap | Kobaz: not an unreasonable request. I'll look into that, too. |
| 14:52.46 | ManxPowerAsteria | I found Grateful Dead Radio. My life is now complete. |
| 14:52.53 | Kobaz | mort_gib: the more the merrier |
| 14:53.06 | Kobaz | mort_gib: nothing like paying 1000 bucks for a single station |
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| 14:53.09 | mort_gib | Thanks :-) I was using the Snom 370, but can see it's not perfect! |
| 14:53.25 | mort_gib | kobaz: Yes, unless YOU have to pay :-) |
| 14:53.40 | ManxPowerAsteria | If YOU have to pay then you don't need 20 line appearances on a sidecar. |
| 14:53.51 | ManxPowerAsteria | If you need that many then you can afford to buy them |
| 14:54.19 | mort_gib | ManxPowerAsteria: Good point! |
| 14:54.21 | Kobaz | and polycom shared line appearances don't even work with asterisk |
| 14:54.34 | ManxPowerAsteria | Kobaz: I've not done SLA, but BLF works just fine. |
| 14:54.35 | Kobaz | you can configure them as blf's or speed dials though |
| 14:54.45 | mort_gib | Kobaz: ?? |
| 14:54.53 | Kobaz | yeah polycom sla isn't supported on asterisk |
| 14:54.55 | mort_gib | "Shared line" huh?? |
| 14:55.07 | Kobaz | mort_gib: shared line... ie: hey joe, pick up on line 2! |
| 14:55.07 | [TK]D-Fender | Kobaz: .... ASTERISK DOESN'T SUPPORT SLA |
| 14:55.09 | ManxPowerAsteria | We didn't really need SLA. One of the programmers where I work patched Asterisk to support several additional busy states. |
| 14:55.23 | Kobaz | [TK]D-Fender: doesn't mean my statement is incorrect |
| 14:55.41 | ManxPowerAsteria | [TK]D-Fender: sla.txt would disagree with you. |
| 14:55.53 | [TK]D-Fender | Kobaz: Sure.. just tack "ANYBODY'S" on there |
| 14:56.14 | [TK]D-Fender | ManxPowerAsteria: that is a FUGLY hack pretending to be SLA, and you know it. |
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| 14:56.44 | Kobaz | i could imagine a hacky way to implement sla |
| 14:56.53 | [TK]D-Fender | Kobaz: We already have it |
| 14:57.13 | mort_gib | Ah, I can't see any justification for SLA... |
| 14:57.18 | Kobaz | move every call into it's own meetme |
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| 14:57.29 | Kobaz | mort_gib: people with old key systems love sla |
| 14:57.44 | mort_gib | SLA seems to be an inherited from old crappy Nortel PBX's |
| 14:57.47 | Kobaz | mort_gib: in small offices... and they are sad when new ip systems don't have it |
| 14:58.02 | mort_gib | Yeah, like *8 wont do the trick |
| 14:59.05 | mort_gib | I haven't heard that comment! I had issues getting call pickup working on the Snoms using the sidecar |
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| 15:02.48 | [TK]D-Fender | Amendment : *'s fake SLA is only HALF the job as well. |
| 15:03.54 | mort_gib | [TK]D-Fender: but what does SLA offer that normal pickup wont?? |
| 15:04.24 | mort_gib | I mean, of course they SEE a line flashing, but the pickup works much better! |
| 15:04.30 | [TK]D-Fender | mort_gib: Pickup works on ringing calls. You can't hijack held calls, etc. Doesn't deal with lines vs phones either IIRC. |
| 15:05.09 | mort_gib | [TK]D-Fender: Ok, true! But there are loads of ways around that! |
| 15:08.34 | mort_gib | It's really funny, you go and implement a completely new infrastructure and you have to copy the limitations, and functions of the existing system.... |
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| 15:09.31 | [TK]D-Fender | *b00m* |
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| 15:12.01 | mort_gib | My VOIP HW pusher is suggesting the Polycom 670 + Sidecar |
| 15:13.07 | [TK]D-Fender | mort_gib: WTF is the need for Gigabit & colour? |
| 15:13.27 | [TK]D-Fender | mort_gib: the 650 I can at least justify the backlight & HD for |
| 15:13.40 | [TK]D-Fender | mort_gib: and "pusher" is right. He's on crack |
| 15:13.53 | pdmmm | haha |
| 15:13.55 | pdmmm | voip pusher |
| 15:13.56 | mort_gib | [TK]D-Fender: Sales pitch :-) |
| 15:14.02 | mort_gib | pusher indeed! |
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| 15:14.09 | pdmmm | how much is that fone? |
| 15:14.35 | [TK]D-Fender | pdmmm: Depends where you buy it. |
| 15:14.49 | pdmmm | average? |
| 15:15.18 | [TK]D-Fender | pdmmm: IP 650 = $266. IP 670 = $355 |
| 15:15.24 | [TK]D-Fender | pdmmm: 100$ waste for nothing |
| 15:15.26 | pdmmm | yikes |
| 15:15.32 | pdmmm | wtf do u get extra? |
| 15:15.47 | [TK]D-Fender | pdmmm: I jsut listed the 2 bigger poitns |
| 15:15.47 | mort_gib | £178 for the IP 650 |
| 15:16.07 | pdmmm | mort_gib: dude, get the 650 |
| 15:16.12 | pdmmm | it sounds cool to have color and shit |
| 15:16.16 | pdmmm | but its just a fone |
| 15:16.21 | pdmmm | its not like an iphone or something :) |
| 15:16.21 | RichardLynch | I'm thinking of extending the Festival() application to take a third parameter "voice" to let you pick which voice it uses. Obviously I'll have to alter the Lisp code in festival.scm Will I also need to alter asterisk C code that does the wrapper calls, or is it automated enough to just pass in whatever extra params are provided? |
| 15:16.37 | mort_gib | and it wont be MY fon anyway :-) |
| 15:17.09 | pdmmm | i bought a 7970 for my living room |
| 15:17.18 | pdmmm | cuz i wanted touch screen and to eventually have home automation |
| 15:17.31 | mort_gib | 7970 is Cisco -Right |
| 15:17.35 | pdmmm | ya |
| 15:17.39 | pdmmm | i like cisco over polycom |
| 15:17.50 | pdmmm | except when they get stupid and only have sccp |
| 15:17.55 | mort_gib | I have issues with Cisco |
| 15:18.43 | [TK]D-Fender | touch-screen... LOL |
| 15:18.54 | [TK]D-Fender | hordes his copy of res_telepathy.so |
| 15:18.59 | pdmmm | what issues w/ cisco? |
| 15:19.05 | pdmmm | [TK]D-Fender: dude, dont be stingy |
| 15:19.13 | pdmmm | hehehe |
| 15:19.23 | mort_gib | I have an issue with a Cisco ADSL router (877) and Cisco wants me to sign up for Smart (for them) net to get an updated IOS |
| 15:19.59 | pdmmm | I can download it for u |
| 15:20.06 | mort_gib | Yeah!! |
| 15:20.14 | pdmmm | i'll msg u my email address |
| 15:20.18 | mort_gib | Damn, I would appreciate that! |
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| 15:22.12 | pdmmm | smartnet is cheap usually anyway |
| 15:22.45 | mort_gib | pdmm: Yes, but my client would have to sign up, not me! |
| 15:22.56 | mort_gib | I don't do that much Cisco work anyway |
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| 15:24.07 | pdmmm | i had polycom 501's a few years ago |
| 15:24.10 | pdmmm | and i just like the cisco better |
| 15:24.47 | mort_gib | i mostly use the Snoms, they are nice, and Users like them better... |
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| 15:25.24 | mort_gib | The website is a BIG selling point! |
| 15:25.31 | Chuggs | mort_gib: how are they to administer |
| 15:26.13 | mort_gib | Chuggs: Great, you lock certain settings when you provision the handsets, users get to "play" which gets some of the resistance out of the way! |
| 15:30.15 | mort_gib | Chuggs: I don't like to mix different phones though! |
| 15:30.55 | fcois93 | I have that error when I try to open a conference : app_meetme.c:837 build_conf: Unable to open pseudo device |
| 15:31.00 | fcois93 | know you why? |
| 15:31.25 | Chuggs | mort_gib: we are just in the planning stages of a new system. we have centrex now <shudder> |
| 15:33.07 | mort_gib | Chuggs: Yes <shudder> depending on the size I would get a few phones to play with! |
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| 15:36.05 | Chuggs | mort_gib: im trying to do that. my Bell guy got me a 7970 to play with but when I found out they where 900+ I wont even put power to it. |
| 15:36.10 | Chuggs | Does this channel have on-line logs ? |
| 15:36.21 | pdmmm | fcois93: dahdi is missing |
| 15:36.27 | pdmmm | aka zaptel |
| 15:36.30 | pdmmm | aka ztdummy |
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| 15:37.06 | mort_gib | Chuggs: LIke I said, I have issues with Cisco! |
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| 15:37.49 | mort_gib | Chuggs: i you buy bulk you get a hefty discount on the Snom handsets, only thing be careful with the firmware! |
| 15:38.41 | fcois93 | pdmmm: how can I conf it? |
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| 15:38.49 | fcois93 | pdmmm: it is install |
| 15:39.26 | pdmmm | modprobe ztdummy |
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| 15:40.19 | fcois93 | pdmmm: FATAL: Module ztdummy not found. |
| 15:40.59 | RichardLynch | fcoic93: Google "install ztdummy" |
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| 15:45.31 | [TK]D-Fender | gah |
| 15:47.08 | fcois93 | RichardLynch: why I need ztdummy to do meetme by SIP ? |
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| 15:47.45 | pdmmm | timing |
| 15:47.45 | ManxPowerAsteria | fcois93: Meetme uses the Zaptel / DAHDI framework for audio mixing and audio timing. You can't run MeetMe without it. |
| 15:47.50 | [TK]D-Fender | fcois93: MeetMe requires a timing source. the channel you throw into it don't matter |
| 15:49.19 | fcois93 | ok |
| 15:51.30 | RichardLynch | I believe meetme has to "sync up" two voice/audio streams. To do that, they need a "common" reference clock/timer. You can buy a fancy hardware clock for high precision. Or use tzdummy for free for "good enough". As I understand it, anyway. |
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| 15:56.55 | doolph | hello, anyone know how many simultaneous channels does this machine can support AMD Duron 1300 with 512MB ram |
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| 15:57.27 | pdmmm | doolph: 21,912 |
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| 15:58.03 | doolph | 21 calls? |
| 15:58.06 | pdmmm | yes |
| 15:58.12 | doolph | how u get that? |
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| 15:58.28 | pdmmm | i dunno |
| 15:58.32 | ManxPowerAsteria | Where I work we usually say that 250 calls is the max per server for use with our software. |
| 15:58.45 | pdmmm | its as variable as you asking how many calls can x machine handle :) |
| 15:59.02 | doolph | but there's any calculator or something |
| 15:59.25 | pdmmm | again |
| 15:59.27 | pdmmm | theres so many factors |
| 15:59.32 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 15:59.32 | doolph | like |
| 15:59.36 | pdmmm | codec |
| 15:59.37 | pdmmm | network |
| 15:59.38 | [TK]D-Fender | doolph: Not with your kind of details |
| 15:59.45 | RichardLynch | Do I have to do anything "special" to get simultaneous calls with VoicePulse?... I tried to use Originate + Dial, both using "primary" and got error noises on the line. Using "primary" for one, and "backup" for the other worked fine... Am I allowed one call per VoicePulse url? Surely not... |
| 15:59.51 | ManxPowerAsteria | doolph: Feel free to write one. Be sure to take into account zaptel .vs. SIP as well as transcoding, codecs, system memory and CPU, as well as HD access speeds, number of simul calls .vs. number of extensions |
| 16:00.12 | pdmmm | theres no generic formula |
| 16:00.15 | pdmmm | is what we're trying to say |
| 16:00.16 | ManxPowerAsteria | RichardLynch: try asking voicepulse. |
| 16:00.32 | pdmmm | thats like joining #windows and asking how many windows can your AMD have open at once? |
| 16:00.39 | doolph | ah ok, its SIP, guess no transcoding, because the provider can support all codecs so the clients too |
| 16:00.49 | ManxPowerAsteria | doolph: no voicemail? |
| 16:01.06 | [TK]D-Fender | ManxPowerAsteria: And don't forget to account for distro chosen, hemispherical compsenation, accound for gravitational and magnetic skew, global warming, and sun-spots. |
| 16:01.15 | doolph | system cpu is amd duron 1300 i think its 800mhz, no voicemail, 512mb ram, hd is an ATA, it has huge internet speed |
| 16:01.21 | ManxPowerAsteria | [TK]D-Fender: none of those really have THAT much of an impact. |
| 16:01.37 | [TK]D-Fender | ManxPowerAsteria: Give or take 100% |
| 16:01.44 | pdmmm | doolph: what are you trying to do? |
| 16:01.46 | ManxPowerAsteria | doolph: assuming all those things I'd put up to 100 people on that server before I start to worry. |
| 16:02.09 | codefreeze-lap | Kobaz: mcnobody: I found it... got it so mcnobody's prob shouldn't happen any more... will commit soon... |
| 16:02.11 | doolph | I am trying to connect a call center |
| 16:02.13 | ManxPowerAsteria | of course with w/W/t/T, call recordings, meetme, features.conf, etc will all impact that. |
| 16:02.28 | doolph | with only 25 people |
| 16:02.31 | [TK]D-Fender | doolph: What do you NEED to support? |
| 16:02.32 | pdmmm | that should be fine |
| 16:02.35 | [TK]D-Fender | doolph: Recording? |
| 16:02.37 | pdmmm | but are u gunna record calls? |
| 16:02.39 | doolph | nothing |
| 16:02.42 | doolph | just make calls |
| 16:02.45 | [TK]D-Fender | doolph: More than fine |
| 16:02.59 | doolph | nice |
| 16:03.07 | pdmmm | there |
| 16:03.09 | pdmmm | answer provided |
| 16:03.11 | pdmmm | next in line please |
| 16:03.16 | doolph | lol |
| 16:03.23 | doolph | ah yes |
| 16:03.24 | doolph | me again |
| 16:03.38 | doolph | is this asterisk 1.6.0.6 is well tested already? |
| 16:03.53 | pdmmm | The Asterisk.org development team is proud to announce the release of |
| 16:03.53 | pdmmm | Asterisk 1.6.0.6. This release is available for download from |
| 16:03.54 | pdmmm | http://downloads.digium.com/. |
| 16:04.03 | [TK]D-Fender | doolph: For at least a week |
| 16:04.33 | pdmmm | i mean |
| 16:04.34 | pdmmm | its asterisk |
| 16:04.36 | pdmmm | not nortel |
| 16:04.39 | stintel | lol |
| 16:04.41 | stintel | good point |
| 16:04.51 | doolph | should I upgrade? |
| 16:04.55 | pdmmm | to nortel? |
| 16:04.58 | doolph | noo |
| 16:04.59 | RichardLynch | LOL |
| 16:05.11 | doolph | i mean from my old 1.4 to 1.6 |
| 16:05.43 | pdmmm | sure |
| 16:06.03 | RichardLynch | Do you need the new features? Are you having problems? Have you tested 1.6 for yourself on your own dev and then staging server properly? Are you using funky hardware? These are all factors we'd have to know to advise you, at best. Not that I'd be able to advise you, personally, being way too new at asterisk. |
| 16:06.16 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.33.119) |
| 16:06.53 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 16:06.54 | doolph | where I can see the new features |
| 16:07.05 | RichardLynch | On the asterisk website? |
| 16:07.28 | RichardLynch | From a link buried in the news release? Just guessing. |
| 16:07.31 | [TK]D-Fender | doolph: CHANGELOG |
| 16:07.33 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 16:07.57 | *** join/#asterisk sasargen (n=chatzill@70-4-5-90.pools.spcsdns.net) |
| 16:08.03 | *** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org) |
| 16:08.26 | *** join/#asterisk msetim (n=setim@200.195.162.210) |
| 16:09.16 | doolph | I don't need dhadi if I don't have any hardware eh |
| 16:11.23 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 16:12.27 | [TK]D-Fender | doolph: MeetMe & IAX Trunk mode require it |
| 16:12.58 | *** join/#asterisk dwery (n=dwery@nslu2-linux/dwery) |
| 16:14.00 | anonymouz666 | Did you know that IAX2 in the past was almost called TASTE? |
| 16:14.02 | anonymouz666 | he he |
| 16:14.32 | tzafrir_laptop | hmm.. so this is why it's called something like "yuck"? |
| 16:16.57 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 16:17.35 | stmaher | hello everyone exten => 705,1,Dial(${BACK_OFFICE}&${RECEPTION},10) used to work with 1.2 why doesnt this work in 1.6? |
| 16:17.49 | stmaher | [Mar 3 16:16:25] WARNING[6803]: app_dial.c:1387 dial_exec_full: Dial argument takes format (technology/[device:]number1) |
| 16:17.53 | fcois93 | I dont understand how to install ztdummy with asterisk 1.6! |
| 16:19.32 | ManxPowerAsteria | stmaher: what does the CLI show for that line |
| 16:19.51 | ManxPowerAsteria | fcois93: download zaptel or dahdi. build. run "make install" and "make config" |
| 16:20.05 | ManxPowerAsteria | then tell your distro to start the zaptel init script installed by "make config". |
| 16:20.40 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 16:20.59 | ManxPowerAsteria | Oh, there are INSTRUCTIONS too. |
| 16:21.11 | stmaher | ManxPowerAsteria SIP/7101-09cc8e58 seems to be the problem where does -09cc8e58 come from? |
| 16:21.23 | stmaher | my variable simply has SIP/7101 in it |
| 16:21.31 | fcois93 | ManxPowerAsteria: FATAL: Module dahdi not found. |
| 16:21.37 | ManxPowerAsteria | stmaher: that is the channel instance number. But you did not provide the information requested. |
| 16:21.47 | ManxPowerAsteria | fcois93: then I guess you should to read the instructions?\ |
| 16:21.54 | stmaher | <PROTECTED> |
| 16:21.54 | stmaher | [Mar 3 16:20:32] WARNING[6921]: app_dial.c:1387 dial_exec_full: Dial argument takes format (technology/[device:]number1) |
| 16:22.14 | fcois93 | ManxPowerAsteria: yes I do, I dont really understand that |
| 16:22.25 | fcois93 | ManxPowerAsteria: what I need to finish it? |
| 16:22.30 | [TK]D-Fender | stmaher: your first value is BALNK |
| 16:22.33 | [TK]D-Fender | BLANK |
| 16:22.35 | ManxPowerAsteria | fcois93: I cannot help you further. |
| 16:23.00 | [TK]D-Fender | fcois93: ztdummy does not exist in 1.6 |
| 16:23.06 | *** part/#asterisk Mog (n=mog@c-68-62-218-186.hsd1.al.comcast.net) |
| 16:23.11 | *** join/#asterisk RobertLaptop (n=rmiddle@pool-72-72-237-239.altnpa.east.verizon.net) |
| 16:23.23 | ManxPowerAsteria | [TK]D-Fender: he knows that since he read the docs. |
| 16:23.41 | fcois93 | [TK]D-Fender: how to do meetme in 1.6 ? |
| 16:23.42 | [TK]D-Fender | ManxPowerAsteria: ... or not :) |
| 16:23.49 | [TK]D-Fender | fcois93: Zaptel NO LONGER EXISTS> |
| 16:23.52 | ManxPowerAsteria | [TK]D-Fender: he said he read the docs. I hope he's not lying. |
| 16:23.57 | [TK]D-Fender | fcois93: Dahdi replaced it months ago |
| 16:24.02 | [TK]D-Fender | ~dahdi |
| 16:24.03 | jbot | [~dahdi] Digium/Asterisk Hardware Device Interface (DAhdi). The new name of zaptel More info at http://www.asterisk.org/zaptel-to-dahdi , and is pronounced "dah-dee" with a short A, or pronounced like http://www.russellbryant.net/dahdi.wav |
| 16:24.15 | ManxPowerAsteria | Lying to someone helping you on IRC can get you blacklisted. |
| 16:24.32 | fcois93 | [TK]D-Fender: I know that dahdi replace zaptel. but how to do meetme now? |
| 16:24.43 | [TK]D-Fender | fcois93: Sames shit, new names |
| 16:24.44 | tzafrir_laptop | dahdi_dummy |
| 16:25.02 | ManxPowerAsteria | that's right just give him the fish and start the fire for him. |
| 16:25.08 | tzafrir_laptop | fcois93, generally: s/zt_/dahdi_/; |
| 16:25.32 | [TK]D-Fender | Generally : s/zt/dahdi_/; |
| 16:26.02 | [TK]D-Fender | ManxPowerAsteria: Only if we can use him as kindling ;) |
| 16:26.18 | ManxPowerAsteria | [TK]D-Fender: I already have in my imagination. |
| 16:26.29 | *** join/#asterisk angryuser (n=gdobrovo@LPuteaux-151-42-35-99.w193-251.abo.wanadoo.fr) |
| 16:26.35 | [TK]D-Fender | Jeff's nuts roasting on an open fire! |
| 16:26.43 | [TK]D-Fender | carols |
| 16:27.23 | *** join/#asterisk docelmo (n=vircuser@pool-151-199-177-240.lyn.east.verizon.net) |
| 16:30.13 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 16:33.02 | *** join/#asterisk De_Mon (i=de_mon@fl-67-77-166-5.dyn.embarqhsd.net) |
| 16:35.03 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 16:38.29 | *** join/#asterisk fexy (n=fexy@208.3.217.29) |
| 16:41.27 | *** join/#asterisk mazpe (n=mazpe@c-71-196-32-43.hsd1.fl.comcast.net) |
| 16:42.27 | mazpe | anyone using g729 on amazon ec2? |
| 16:43.23 | *** join/#asterisk sjobeck (n=sjobeck@208-151-246-203.dq1sn.easystreet.com) |
| 16:44.11 | keith4 | that's an interesting question |
| 16:44.50 | fcois93 | I dont find how to use meetme in 1.6 ! |
| 16:45.27 | Qwell | fcois93: core show application meetme |
| 16:46.06 | *** join/#asterisk talntid (n=eric@c-67-185-19-53.hsd1.wa.comcast.net) |
| 16:46.17 | mazpe | keith4: i'm not even quite sure which CPU to use... http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.6.0/x86-32/ |
| 16:46.25 | keith4 | fcois93: seriously? they just told you |
| 16:46.44 | mazpe | since its a xen virtual machine |
| 16:46.44 | kannan | any one with experience with t38modem + asterisk + hylafax? |
| 16:47.13 | *** join/#asterisk HermesNeto (i=HermesNe@200.249.176.44) |
| 16:47.13 | keith4 | mazpe: do you have a VM from EC2 already? |
| 16:47.23 | mazpe | yeah, its already running asterisk |
| 16:47.30 | keith4 | what does uname -a say? |
| 16:47.52 | mazpe | Linux domU-12-31-38-00-25-44 2.6.18-xenU-ec2-v1.0 #2 SMP Tue Feb 19 10:51:53 EST 2008 i686 athlon i386 GNU/Linux |
| 16:47.53 | mazpe | duh |
| 16:47.54 | mazpe | athlon |
| 16:48.05 | talntid | [Mar 3 08:47:05] WARNING[20002]: file.c:871 ast_streamfile: Unable to open /etc/asterisk/sound/thankyouforcalling (format 0x4 (ulaw)): No such file or directory |
| 16:48.05 | talntid | [Mar 3 08:47:05] WARNING[20002]: pbx.c:5714 pbx_builtin_background: ast_streamfile failed on SIP/10.21.0.22-00920550 for /etc/asterisk/sound/thankyouforcalling |
| 16:48.13 | talntid | that file is there..... any ideas on this? |
| 16:48.28 | mazpe | keith4: i'm asuming the athlon.. and not the athlon xp |
| 16:48.57 | keith4 | mazpe: dunno. looks like it's i686 compatible, no? |
| 16:49.14 | mazpe | not sure the difference |
| 16:50.35 | [TK]D-Fender | talntid: Show us <- |
| 16:50.54 | talntid | Show you what? |
| 16:51.06 | RichardLynch | talntid: It's looking for a ulaw file. Do you have ulaw or some other format?... |
| 16:51.17 | talntid | it was .gsm |
| 16:51.18 | [TK]D-Fender | talntid: an "ls -l" dump confirming the state and location of the file. |
| 16:51.42 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 16:51.48 | talntid | root@pbx2:/etc/asterisk/sound# ls -l |
| 16:51.48 | talntid | total 1840 |
| 16:51.48 | talntid | -rw-r--r-- 1 root root 41217 Apr 27 2008 thankyouforcalling.gsm |
| 16:51.59 | fcois93 | keith4: I know that I have to instal dahi_dummy |
| 16:52.11 | *** join/#asterisk Badrobot- (i=BadRobot@cpe-76-173-233-75.socal.res.rr.com) |
| 16:52.11 | [TK]D-Fender | talntid: And who is * running as? |
| 16:52.34 | talntid | running as "asterisk" |
| 16:52.43 | [TK]D-Fender | talntid: them maybe you should change the OWNER |
| 16:52.59 | Qwell | $20 says the dir isn't o+x |
| 16:53.01 | ManxPowerAsteria | talntid: what is the output of "type /etc/asterisk/sound/thankyouforcalling.*" |
| 16:53.05 | Qwell | any takers? |
| 16:53.14 | [TK]D-Fender | Qwell: I'll take your $20 |
| 16:53.21 | [TK]D-Fender | Qwell: You can keep the bet ;) |
| 16:53.28 | RichardLynch | Can I get half of that? |
| 16:53.29 | Qwell | talntid: ls -ld /etc/asterisk/sounds/ |
| 16:53.37 | Qwell | erm, -s |
| 16:54.12 | talntid | changing owner kept same issue |
| 16:54.27 | Qwell | show me what that command says |
| 16:54.28 | *** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman) |
| 16:54.50 | talntid | ManxPowerAsteria: type: not found |
| 16:55.03 | Qwell | ManxPowerAsteria: type is dos :p |
| 16:55.05 | ManxPowerAsteria | talntid: might be "file" instead of "type" |
| 16:55.07 | Qwell | file is what you're looking for |
| 16:55.12 | talntid | root@pbx2:/etc/asterisk/sound# ls -s /etc/asterisk/sound/ |
| 16:55.12 | talntid | total 1840 |
| 16:55.12 | talntid | <PROTECTED> |
| 16:55.20 | Qwell | talntid: That isn't what I said to type. |
| 16:55.42 | talntid | did you mean -lds? |
| 16:55.45 | ManxPowerAsteria | talntid: thanks for leaving out the flags on that ls that would have provided the information Qwell was looking for. |
| 16:55.47 | Qwell | no, I meant -ld |
| 16:56.04 | talntid | oh, |
| 16:56.11 | talntid | your -s was to remove the s from the sound. ;P |
| 16:56.18 | talntid | i thought you were providing different flags. |
| 16:56.31 | Qwell | the suspense is killing me. |
| 16:56.32 | talntid | drwxr-xr-x 2 root root 4096 Mar 2 20:57 /etc/asterisk/sound/ |
| 16:56.47 | talntid | it should be noted, that this all worked before. |
| 16:56.52 | talntid | until i tried changing the greeting |
| 16:56.54 | fcois93 | I compiled dahdi and I have when I start it : Loading DAHDI hardware modules: FATAL: Module dahdi not found. why? |
| 16:56.55 | Qwell | ...before what? |
| 16:57.03 | Qwell | fcois93: because you didn't install it |
| 16:57.12 | ManxPowerAsteria | I suspect the .gsm file is not 8Khz or not mono or not 16-bit. |
| 16:57.21 | talntid | i created a new .gsm file, renamed the old one, named the new one the same as the old one... and tried it. no good. |
| 16:57.28 | talntid | so i renamed the old one back, and the issue still remained |
| 16:57.33 | Qwell | 41217...should be a multiple of 33 |
| 16:57.49 | Qwell | and it is |
| 16:57.52 | Qwell | which means it's fine |
| 16:58.29 | Qwell | 25 seconds long? |
| 16:58.33 | fcois93 | Qwell: I have no error in my compil ! |
| 16:58.53 | ManxPowerAsteria | Qwell: fcois93 is having ID ten T problems. |
| 16:59.14 | Talkradio | heh |
| 16:59.38 | *** join/#asterisk unasi7 (n=unasi7@194.230.146.72) |
| 16:59.47 | talntid | heh |
| 17:01.01 | Qwell | talntid: are format_gsm.so and codec_gsm.so loaded? |
| 17:01.06 | Qwell | and codec_ulaw.so |
| 17:02.25 | talntid | yes, according to modules show |
| 17:02.51 | pdmmm | fcois93: you dont have dahdi |
| 17:02.51 | pdmmm | http://www.asterisk.org/node/48540 |
| 17:02.51 | pdmmm | install that |
| 17:03.08 | pdmmm | if in your asterisk source directory u do make menuconfig |
| 17:03.16 | pdmmm | and chan_dahdi is XXX'd out |
| 17:03.23 | pdmmm | then do a ./configure again |
| 17:03.28 | Qwell | talntid: show me. pastebin |
| 17:03.32 | fcois93 | ok thank you |
| 17:04.58 | talntid | http://pastebin.com/m338a5c58 |
| 17:05.20 | talntid | Qwell: would you like to ssh into my laptop and screen into the server? |
| 17:06.05 | Qwell | tons of out-of-tree modules.. can't help you |
| 17:06.43 | talntid | hmm. |
| 17:06.53 | *** join/#asterisk tamiel (n=tamiel@ip-176.net-89-2-117.rev.numericable.fr) |
| 17:07.31 | Qwell | core show translations - do you get a conversion time for gsm>ulaw? |
| 17:08.51 | talntid | 2 |
| 17:09.45 | talntid | http://pastebin.com/m6e8e691c |
| 17:10.47 | Qwell | shrug |
| 17:10.51 | pdmmm | its prolly something gay like selinux or something |
| 17:11.05 | mazpe | on 1.6.0.5 is there a command to see all the install codec? |
| 17:11.43 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 17:13.03 | talntid | grr |
| 17:13.05 | talntid | found the issue |
| 17:13.32 | *** join/#asterisk j_o_e (n=silas@ool-18b9f833.dyn.optonline.net) |
| 17:13.55 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 17:14.04 | talntid | but would have been hard for you to diagnose. |
| 17:14.18 | talntid | i mentioned that i renamed the old file, replaced with new file.. |
| 17:14.31 | talntid | well, i renamed it to.. thankyouforcalling.old.gsm |
| 17:14.40 | talntid | so that left 2 files named thankyouforcalling.* |
| 17:15.07 | talntid | so i renamed all the files, and it worked. |
| 17:15.10 | Qwell | .old.gsm isn't a valid extension |
| 17:15.19 | j_o_e | hello, I have a budgetone 200 hardphone. It's able to receive calls and everything works perfectly. However, when I dial out voices are not audible and all I hear on the other end is fast clicking/popping noises (sort of like the predator) ... How can I troubleshoot this? My theory is that it's vocoder related and I tried unsuccessfully to force ulaw |
| 17:17.37 | tamiel | hello, I have two TE412P cards on the same server, do I need to set one timing source per card (currently, card 2 is set to Sync source : Internally clocked) ? |
| 17:17.51 | tamiel | my system.conf (dahdi) : http://asterisk.pastebin.ca/1352041 |
| 17:18.06 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 17:19.53 | Qwell | tamiel: I do believe so |
| 17:20.46 | RichardLynch | I would expect the cards to need to share a timing source, if you expect them to do anything syncrhonized... |
| 17:21.05 | RichardLynch | But I have no real idea... |
| 17:21.17 | Qwell | RichardLynch: sounds right |
| 17:21.30 | Qwell | but - I don't ultimately know either :p |
| 17:26.21 | *** join/#asterisk docid (n=eris@69.196.68.142) |
| 17:28.00 | tamiel | I use dahdi_genconf and it configures system.conf like that. |
| 17:28.31 | tzafrir_laptop | tamiel, sadly there's some misunderstanding here between dahdi_genconf and wct4xxp: |
| 17:28.42 | tzafrir_laptop | span=4,4,0,ccs,hdb3 (that is: timing=4) |
| 17:28.50 | fcois93 | pdmmm: I don't have chan_dahdi when I do ./configure in asterisk source folder |
| 17:29.08 | tzafrir_laptop | for wct4xxp, any value of timing >=4 equals to 0 |
| 17:29.14 | [TK]D-Fender | fcois93: Show us you've installed and configured DAHDI |
| 17:29.28 | docid | umm, what does the hdb3 mean? ive been trying to configure a sangoma a102 for mfc/r2 signaling, |
| 17:29.48 | tzafrir_laptop | docid, you need cas, then |
| 17:30.02 | fcois93 | [TK]D-Fender: I juste have done make && make install && make config |
| 17:30.06 | tzafrir_laptop | (look for 'R2' in genconf_parameters) |
| 17:30.11 | fcois93 | [TK]D-Fender: I dont know what to do more |
| 17:30.19 | docid | yes, i know, but all the turtorials mention hdb3 after cas |
| 17:30.23 | tamiel | tzafrir_laptop: ok so I need to put all others to 2 (span 2 to 8) ? |
| 17:30.24 | [TK]D-Fender | fcois93: INITIALIZE the damn thing and SHOW US |
| 17:30.38 | docid | what is that paramater? |
| 17:30.42 | *** join/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 17:30.49 | tzafrir_laptop | for building asterisk, it only needs to be installed |
| 17:30.58 | docid | every time i change settings to reflect r2, dahdi quits working |
| 17:31.16 | tzafrir_laptop | fcois93, egrep -i 'dahdi|tonezone' buildtools/menuselect-deps |
| 17:31.32 | [T]ank | anyone here located in idaho? I was contacted by someone from there to build a phone system for them. They are too far away for me and I am looking for someone to refer them to. |
| 17:31.37 | docid | but i did build it with the open r2 lib, and ldd shows its there |
| 17:31.53 | tzafrir_laptop | docid, which r2 do you use? unicall or openr2? |
| 17:32.04 | tzafrir_laptop | openr2 |
| 17:32.04 | docid | trying to use openr2 |
| 17:32.16 | fcois93 | tzafrir_laptop: egrep -i 'dahdi|tonezone' buildtools/menuselect-deps grep: buildtools/menuselect-deps: Aucun fichier ou répertoire de ce type |
| 17:32.51 | tzafrir_laptop | fcois93, egrep -i 'dahdi|tonezone' build_tools/menuselect-deps |
| 17:32.57 | tzafrir_laptop | my typo |
| 17:33.10 | docid | except im not sure if t1 robbed bit cas is the same thing as mfc/r2, were in the artic, apperently the last switch canada that uses this signaling |
| 17:33.13 | [TK]D-Fender | fcois93: and what does dahdi_cfg -vvv have to say? did you modprobe the proper driver? |
| 17:33.26 | fcois93 | http://pastebin.com/m190f025d |
| 17:33.53 | fcois93 | DAHDI Tools Version - 2.1.0.2 |
| 17:33.57 | [TK]D-Fender | fcois93: Did you reboot afterwards? |
| 17:34.00 | fcois93 | line 0: Unable to open master device '/dev/dahdi/ctl' |
| 17:34.10 | fcois93 | [TK]D-Fender: no, I didnt reboot |
| 17:34.17 | fcois93 | I have to? |
| 17:34.19 | coppice | docid: something in canada uses R2? :-\ |
| 17:34.31 | tzafrir_laptop | fcois93, that's an unrelated error |
| 17:34.37 | docid | yeah, up inside the arctic circle |
| 17:34.41 | [TK]D-Fender | fcois93: modprobe dahdi and dahdi_dummy, reboot, and redo "dahdi_cfg -vvvv |
| 17:34.42 | tzafrir_laptop | what's the output of that command I asked you |
| 17:34.56 | tzafrir_laptop | and no: no real need to reboot |
| 17:35.06 | tzafrir_laptop | just a 'modprobe dahdi_dummy' |
| 17:35.18 | tzafrir_laptop | but this is irrelevant for *building* asterisk |
| 17:35.19 | *** join/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 17:35.24 | fcois93 | FATAL: Module dahdi_dummy not found |
| 17:35.32 | fcois93 | idem for dahdi |
| 17:35.40 | coppice | docid: are you sure its R2? canada is all T1 territory, and nobody ever used R2 on T1s |
| 17:35.48 | tzafrir_laptop | fcois93, again, I asked you for some output |
| 17:36.04 | docid | well, our provider clams its mfc-r2 itu on t1 |
| 17:36.13 | tzafrir_laptop | coppice, is there anything inherent for E1 in MFC/R2 ? |
| 17:36.39 | docid | but our other equipment shows t1 robbed bit cas, im not sure if those are synonamous |
| 17:37.02 | fcois93 | tzafrir_laptop: egrep -i 'dahdi|tonezone' buildtools/menuselect-deps grep: buildtools/menuselect-deps: Aucun fichier ou répertoire de ce type |
| 17:37.25 | tzafrir_laptop | R2 is higher level, and lower-level is indeed CAS with robbed bit signalling |
| 17:37.34 | coppice | the line signalling for R2 is based on the ABCD bits of an E1 in CAS mode. However, R2 only really uses 2 of the bits, so it could be handled by the T1 A and B robbed bits |
| 17:38.16 | tzafrir_laptop | fcois93, egrep -i 'dahdi|tonezone' build_tools/menuselect-deps |
| 17:38.20 | coppice | tzafrir_laptop: E1's don't rob any bits for CAS :-) |
| 17:38.28 | tzafrir_laptop | right |
| 17:38.32 | docid | ahhh, interesting, so, the hdb3 paramater, surrently ours are AMI (the utstarcom hooked up to simular repots this as the signaling that works) |
| 17:38.39 | *** join/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 17:38.49 | tzafrir_laptop | Still Zaptel calls it that way, which confused me |
| 17:39.01 | coppice | HDB3 is an E1 parameter. it is not applicable to T1s |
| 17:39.14 | docid | ahh, that could be part of whats throwin me |
| 17:39.36 | docid | thank you....i think today is gunna be spent chatting with provider and sangoma possibly |
| 17:40.11 | fcois93 | I just reboot and nothings change |
| 17:40.49 | tzafrir_laptop | whistels along |
| 17:40.56 | fcois93 | [TK]D-Fender: FATAL: Module dahdi not found. after reboot |
| 17:41.10 | tzafrir_laptop | surprise, surprise |
| 17:42.21 | fcois93 | any idea? |
| 17:42.30 | tzafrir_laptop | fcois93, egrep -i 'dahdi|tonezone' build_tools/menuselect-deps |
| 17:42.51 | *** part/#asterisk dlewis (i=c7340d67@about/security/staff/dlewis) |
| 17:43.19 | tzafrir_laptop | nice to have history in the IRC client. It saves me typing |
| 17:43.22 | fcois93 | tzafrir_laptop: grep: build_tools/menuselect-deps: Aucun fichier ou répertoire de ce type |
| 17:43.34 | *** join/#asterisk Ericounet (n=Ericoune@ACaen-151-1-18-240.w86-215.abo.wanadoo.fr) |
| 17:43.39 | tzafrir_laptop | in the asterisk source directory? |
| 17:43.55 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-18-240.w86-215.abo.wanadoo.fr) |
| 17:44.09 | *** join/#asterisk apocn (n=apo@unaffiliated/apocn) |
| 17:44.20 | fcois93 | tzafrir_laptop: DAHDI=1 TONEZONE=1 |
| 17:44.35 | apocn | Hello, is there a way to move a client from one queue to the other from the CLI or Manager Interface? |
| 17:45.16 | [TK]D-Fender | apocn: AMI Redirect |
| 17:45.24 | *** part/#asterisk [T]ank (n=ckwall@206.71.78.158) |
| 17:45.55 | tzafrir_laptop | fcois93, in the dahdi source directory, do you have: drivers/dahdi/dahdi_dummy.ko #? |
| 17:46.03 | apocn | [TK]D-Fender, thanks |
| 17:46.49 | fcois93 | tzafrir_laptop: yes I have it |
| 17:47.57 | tzafrir_laptop | next: does the following command return plenty of output or an error? modinfo dahdi_dummy |
| 17:48.21 | fcois93 | tzafrir_laptop: modinfo: could not find module dahdi_dummy |
| 17:48.31 | Qwell | didn't run make install... |
| 17:48.44 | fcois93 | I run make instal!!! |
| 17:48.55 | tzafrir_laptop | was there any error in it? |
| 17:49.05 | fcois93 | I have all in init.d/ ... |
| 17:49.10 | tzafrir_laptop | make install && echo "all was well" |
| 17:49.11 | fcois93 | no error in make install... |
| 17:49.18 | tzafrir_laptop | (in dahdi-linux) |
| 17:50.01 | pdmmm | ugh |
| 17:50.03 | pdmmm | still dahdi crap?! |
| 17:50.21 | fcois93 | all was well |
| 17:50.28 | tzafrir_laptop | fcois93, so repeat that command: |
| 17:50.33 | tzafrir_laptop | next: does the following command return plenty of output or an error? modinfo dahdi_dummy |
| 17:50.50 | fcois93 | modinfo: could not find module dahdi_dummy |
| 17:51.21 | Qwell | ~pastebin |
| 17:51.22 | jbot | [~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste |
| 17:51.27 | Qwell | fcois93: show us the output of make install |
| 17:51.33 | *** join/#asterisk j_kroon (n=jkroon@dsl-240-144-29.telkomadsl.co.za) |
| 17:51.41 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 17:52.09 | fcois93 | http://pastebin.com/d153c8908 |
| 17:52.38 | *** join/#asterisk Gido-E (n=gido@lounge.datux.nl) |
| 17:53.51 | tzafrir_laptop | hmm... non matching versions in lines 37 and 39 |
| 17:54.03 | tzafrir_laptop | fcois93, what's the output of: uname -r |
| 17:54.08 | Qwell | odd |
| 17:54.28 | fcois93 | 2.6.18-6-686 |
| 17:54.43 | docid | why does wanpipe still look for dev/zap when its installed for dahdi, i do have a /dev/dahdi i dont have a dev zap |
| 17:54.52 | tzafrir_laptop | ls -ld /lib/modules/2.6.18-6-686/build /usr/src/linux |
| 17:54.53 | Qwell | docid: Ask Sangoma.. |
| 17:54.55 | docid | will just makeing a softlink work? |
| 17:55.03 | docid | ahh |
| 17:55.05 | docid | good point |
| 17:55.08 | RichardLynch | docid: I dunno, but ln -s /dev/dahdi /dev/zap would prbably fix it. |
| 17:55.19 | fcois93 | lrwxrwxrwx 1 root src 36 2008-11-03 17:29 /usr/src/linux -> /usr/src/linux-headers-2.6.18-5-686/ |
| 17:56.02 | tzafrir_laptop | Note, however, that zaptel and dahdi have incompatible ABIs. If a program was built to use a zaptel ioctl it will fail to use dahdi (kernel) |
| 17:56.15 | docid | aye, that idnt werk |
| 17:56.18 | *** part/#asterisk mercutioviz (n=chatzill@freeswitch/developer/msc) |
| 17:56.23 | tzafrir_laptop | fcois93, aptitude install linux-headers-`uname -r` |
| 17:56.29 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 17:56.46 | *** join/#asterisk mellow-yellow (n=mellow-y@mycomp.norris-stevens.com) |
| 17:57.27 | fcois93 | I have to retry to make && make install ? |
| 17:58.26 | fcois93 | modinfo dahdi_dummy find! |
| 17:58.28 | *** join/#asterisk paulproteus (n=paulprot@2002:db69:2513:0:0:0:0:1) |
| 17:58.41 | fcois93 | tzafrir_laptop: Thank you so much! |
| 17:59.15 | fcois93 | it is 7pm in Paris, end of my day. a beautiful end! |
| 17:59.22 | fcois93 | thank you tzafrir_laptop! |
| 17:59.53 | fcois93 | I think that the next operations are to recompil asterisk... |
| 18:02.39 | Gido-E | which gcc is used for compiling the g729 codecs? |
| 18:03.01 | Qwell | Gido-E: 4.3 I believe |
| 18:04.02 | Gido-E | is it possible, that somebody recompiles it with 3.3.4? |
| 18:04.14 | Qwell | Gido-E: It shouldn't matter at all... |
| 18:05.22 | Gido-E | Qwell i know. But... |
| 18:05.27 | Qwell | but? |
| 18:05.39 | codefreeze-lap | Kobaz: been thinking about the optimization you suggested for AEL earlier, dumping the $[] if it's a constant... but ${} is not all that is required to make it variable.... |
| 18:06.34 | Gido-E | this morning i installed the g729 codec on a machine. with a 50 user license. and today it behaved verry strainge |
| 18:07.06 | Gido-E | only about 200 sip accounts. on a 8 core machine. |
| 18:07.22 | Qwell | "behaved strange" doesn't tell me anything.. |
| 18:09.58 | Kobaz | codefreeze-lap: mmm |
| 18:10.07 | Kobaz | codefreeze-lap: what else can be a variable |
| 18:11.24 | Kobaz | Gido-E: i went to start my car this morning, and it was very strange |
| 18:11.33 | *** join/#asterisk ZcCOP (n=dave@204.97.128.131) |
| 18:11.34 | Kobaz | Gido-E: can you help me fix it? |
| 18:11.53 | ZcCOP | I need some assistance in dialing across a cluster of asterisk servers.. (IAX) |
| 18:12.10 | Kobaz | ZcCOP: more details? |
| 18:12.13 | Gido-E | Kobaz :-) |
| 18:12.24 | ZcCOP | i have a end point registered to server C, which is in this case, trixbox that has many DID associated with it. |
| 18:12.35 | ZcCOP | the call will arrive on server A, to for example, 5551212. |
| 18:12.47 | ZcCOP | I need to be able to call 5551212 on the SIP account registered with server C |
| 18:13.21 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 18:13.35 | ZcCOP | I can get as far as passing the call to the correct SIP account on server C, but by the time i get that far, I lost the ability to call a certain extension/DID on that sip account |
| 18:13.57 | Kobaz | ZcCOP: so can't you just send the Dial() over to the sip peer you need |
| 18:14.23 | *** join/#asterisk SkywaIker (n=pirch@58.147.17.166) |
| 18:14.23 | ZcCOP | no, because that sip peer is only registered with server c, and wont accept a call from server a directly |
| 18:14.24 | Kobaz | Dial(SIP/foo/5551234) |
| 18:14.44 | ZcCOP | all the calls come into PRI's assoicated with server A. |
| 18:14.58 | Kobaz | what kind of mishmosh do you have? |
| 18:15.02 | Kobaz | pri to iax to sip? |
| 18:15.03 | ZcCOP | so I need that call to somehow get to server C, and ultimately to 5551212@sippeer |
| 18:15.18 | ZcCOP | thats what I was trying.. but im open to do whatever is needed |
| 18:15.27 | Kobaz | what about pri to sip? |
| 18:15.46 | ZcCOP | DUNDi was my first idea, but that doesnt allow me to easily point MANY did's to a sippeer registered with a different server |
| 18:15.50 | Kobaz | dial(SIP/sippeer/5551212) |
| 18:16.15 | ZcCOP | problem is, the sippeer is registered to Server C, not Server A |
| 18:16.30 | j_kroon | hi all, when "pulling" calls using *8, I'd like to have the option of doing some call recording checks and potentially activating call recording before bridging the call. |
| 18:16.36 | j_kroon | what are my options, if any? |
| 18:16.36 | ZcCOP | the PRI's are directly connected to Server A |
| 18:16.38 | Kobaz | a sip peer doesn't care what number it gets as a dnid |
| 18:16.52 | Kobaz | you can send it anything, including ascii |
| 18:17.21 | Kobaz | j_kroon: you can do a custom feature code that calls an agi script |
| 18:17.23 | ZcCOP | but then I neeed to have trixbox (my end user) allow incoming SIP from all my boxex, right? |
| 18:17.38 | Kobaz | ZcCOP: you can restict by host if you want |
| 18:17.51 | ZcCOP | im trying to do this as scalable as I can |
| 18:18.26 | Kobaz | you'll have to detail more what you're doing... i don't see any problems with what you've said so far |
| 18:18.32 | j_kroon | Kobaz, can i switch the feature code off, and have *8 as a dialplan app that just ends with a NoCDR(); Pickup() sequence? |
| 18:18.46 | Kobaz | j_kroon: yeah |
| 18:18.47 | ZcCOP | right now i have 3 Asterisk servers, A has PRI for all inbound origination, Servers B and C will serve as SIP registration servers |
| 18:19.09 | j_kroon | is it possible to add channel variables to the channel being picked up? |
| 18:19.24 | ZcCOP | so ideally, a trixbox can register to servers B and C, so if one or the other goes down, service wont be effected |
| 18:19.30 | j_kroon | doesn't look like it ... no :( |
| 18:19.35 | Kobaz | j_kroon: see the docs for res_features.conf, and there's some stuff about making custom feature codes, and you have to turn them on, on a per-call basis with DYNAMIC_FEATURES |
| 18:19.52 | j_kroon | Kobaz, thanks. |
| 18:19.57 | Kobaz | j_kroon: yes you can, see Dial's M option |
| 18:20.22 | j_kroon | Kobaz, I don't Dial() do I? I end with Pickup(). |
| 18:20.25 | ZcCOP | Kobaz, im trying to setup a cluster senerio, where my customers can regisisted to any one of my servers, and have calls still arrive to their gear |
| 18:20.38 | Kobaz | ZcCOP: yeah |
| 18:21.00 | Kobaz | ZcCOP: so on the registration servers, send the calls over sip back to the pri server |
| 18:21.04 | Kobaz | and vise versa |
| 18:21.13 | j_kroon | ie, one channel (the one being picked up) goes into Dial(SIP/foo) and then the other channel gets into Pickup() .. which then bridges the two channels into a call? |
| 18:21.14 | *** part/#asterisk ManxPowerAsteria (n=Administ@router.asteriasgi.com) |
| 18:21.16 | ZcCOP | thats fine, i have outbound calling functioning.. |
| 18:21.32 | ZcCOP | the problem is that one of my trixbox customers, can have 1 or 300 DID's routed to them |
| 18:21.40 | Kobaz | yeap |
| 18:21.43 | Kobaz | and that's fine |
| 18:21.49 | ZcCOP | and that needs to somehow flow over from PRI server, to whatever reg server their on |
| 18:22.10 | ZcCOP | and still dial(SIP/thedid@mycustomer) |
| 18:22.16 | Kobaz | okay |
| 18:22.17 | ZcCOP | im using realtime, btw |
| 18:22.17 | Kobaz | so |
| 18:22.23 | Kobaz | you need to run some code in the background |
| 18:22.30 | *** join/#asterisk jpcansa (n=jpbenavi@201.203.102.70) |
| 18:22.45 | Kobaz | so now i see the problem... the inbound pri call doesn't know which reg server to send it to |
| 18:23.14 | Kobaz | when the client registers, there is an event you can get via AMI |
| 18:23.15 | ZcCOP | well i actually got that going ok, but once it sends the call to the right reg server, the reg server doesnt know how to send the call DNID to the sip-friend |
| 18:23.37 | Kobaz | and then you can update the pri server with the reg server to go to |
| 18:23.48 | ZcCOP | i use Realtime() to lookup what regserver a particular sip-friend is registered to |
| 18:23.52 | jpcansa | does anybody know why my asterisk is crashing like this:--- http://pastebin.com/fa52d2e7 ----I think it is happening since i upgraded to 1.4.23.1. any aidea???? |
| 18:23.59 | Kobaz | that works too |
| 18:24.03 | Kobaz | okay so |
| 18:24.41 | Kobaz | ${EXTEN} is going to be the dnid |
| 18:25.15 | ZcCOP | right |
| 18:25.31 | Kobaz | so then in your context for the sip user |
| 18:25.41 | Qwell | Gido-E: no explanation? |
| 18:25.43 | ZcCOP | So, heres my dial command at PRI server, SIP/${m_regserver}/customersipaccountname/${EXTEN}|120 |
| 18:25.47 | Kobaz | you need to pass that ${EXTEN} to Dial(to your sip customer) |
| 18:26.02 | Kobaz | k that looks good |
| 18:26.16 | ZcCOP | on the reg server, i get Call from 'alb-sip1' to extension 'customersipaccount/5186893652' rejected |
| 18:26.29 | ZcCOP | (that number is a test numner, fyi) |
| 18:26.40 | Kobaz | 518, heh |
| 18:26.42 | Kobaz | albany? |
| 18:26.56 | ZcCOP | yessir |
| 18:27.04 | Kobaz | maybe i should just stop by :P |
| 18:27.12 | Kobaz | anyways |
| 18:27.14 | ZcCOP | lol |
| 18:27.47 | Kobaz | so here's the problem |
| 18:27.51 | Kobaz | SIP/${m_regserver}/customersipaccountname/${EXTEN} |
| 18:28.01 | Kobaz | your sending the dnid as customersipaccountname/${EXTEN} |
| 18:28.07 | Gido-E | Qwell, sorry. At this moment not. |
| 18:28.19 | Kobaz | you need to match on that |
| 18:28.24 | Kobaz | and then handle the call accordingly |
| 18:28.44 | ZcCOP | so theres no way to pass a call over SIP to a specific context then? |
| 18:28.53 | ZcCOP | its all dependant on the sippeers context? |
| 18:29.07 | Kobaz | you can't send it to a context that's not defined in the sip peer |
| 18:29.12 | Kobaz | that would be a huge security hole |
| 18:29.25 | ZcCOP | figured, i knew this, im just getting burnt out playing around with it |
| 18:29.28 | Kobaz | you have to use the context that the sip peer is assigned |
| 18:29.37 | Kobaz | sooooo |
| 18:29.39 | Kobaz | what you can do |
| 18:29.45 | Kobaz | make it easy, and flip it around |
| 18:29.56 | Kobaz | SIP/${m_regserver}/${EXTEN}@customersipaccountname |
| 18:30.09 | Kobaz | in your context, match on like... _X! |
| 18:30.17 | Kobaz | and then use CUT to split on the @ |
| 18:30.34 | Kobaz | and then send it to the appropriate sip peer/${EXTEN} |
| 18:31.34 | ZcCOP | interesting. |
| 18:31.42 | ZcCOP | never thought about CUT parsing anything |
| 18:31.45 | ZcCOP | hmmz |
| 18:32.04 | Kobaz | i use it all the time |
| 18:32.51 | Kobaz | this is all fine and dandy assuming your customers are all dialing numeric destinations |
| 18:33.02 | Kobaz | and you're not supporing dialing sip:joe@foo.com |
| 18:33.05 | ZcCOP | i was hyped about using DUNDi for all of this, but it ended up (so I think anyways) useless for passing calls to sipfriends that need to answer for many DIDs |
| 18:33.27 | *** join/#asterisk bbryant (n=brett@68.208.65.50) |
| 18:33.33 | ZcCOP | yea this is all digit based dialing |
| 18:33.47 | Kobaz | dundi is for having a loose network of peers and using your peers to get to their peers |
| 18:34.07 | Kobaz | ie: i have no idea where to send this call, maybe peer A knows... |
| 18:34.09 | j_kroon | ZcCOP, why? I used DUNDi for those kinds of purposes before. |
| 18:34.15 | j_kroon | slightly abused, but it worked. |
| 18:34.23 | NovceGuru | sdfsaf |
| 18:34.30 | Kobaz | yeah, i could imagine it being used, bu it's not the purpose of it, heh |
| 18:34.35 | j_kroon | always wanted to give ENUM a shot too but didn't manage to figure that one out. |
| 18:35.01 | ZcCOP | my goal here was to simply provide multiple servers a peer could register to, and still get calls |
| 18:35.06 | Kobaz | yeah |
| 18:35.31 | j_kroon | Kobaz, which is what made it so damn hard in my case. there was simply no docs on doing exactly what i wanted, and only once I started understanding it better did I actually get it to do what I wanted (which is an ENUM type of thing) |
| 18:35.34 | Kobaz | so what do you do, slap together a bunch of pri's and sell minutes over sip? |
| 18:36.02 | j_kroon | one could :). in effect. |
| 18:36.43 | j_kroon | no, just wanted to advertise my clients's numbers to my other clients so that they can know to call those calls over voip instead of over their own PRIs. |
| 18:37.10 | Kobaz | ah |
| 18:37.46 | Shaun2222 | any good stats utils for asterisk... Somthing that show's stats about incomming/outgoing calls, queue information, staff call details, etc.. |
| 18:37.50 | ZcCOP | thats one way to force your customers to talk :) |
| 18:38.06 | Kobaz | Shaun2222: not really |
| 18:38.33 | ZcCOP | Kobaz: sanity check.. Executing Dial("Zap/73-1", "SIP/regserver3/5555551212@sipfriend") Unable to create channel of type 'SIP' (cause 20 - Unknown) |
| 18:38.58 | ZcCOP | sip debug on regserver3 shows nothing |
| 18:39.14 | [TK]D-Fender | ZcCOP: Because the call is going to "sipfriend" |
| 18:39.18 | *** join/#asterisk wtca (n=wtca@dhcp-93.noc.sonic.net) |
| 18:39.21 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 18:39.27 | Kobaz | oooh, whoops |
| 18:39.48 | ZcCOP | ok.. so SIP/numer@sipfriend@regserver3? |
| 18:39.49 | Kobaz | do umm |
| 18:39.57 | nullable_type | Do you guys know why "show channels" doesnt work in CLI? I want to hangup a call to a number that is already in place |
| 18:39.59 | Kobaz | well |
| 18:40.09 | Kobaz | you wont want the double @ |
| 18:40.11 | *** join/#asterisk Rabenklaue (n=Rabe@g227161071.adsl.alicedsl.de) |
| 18:40.18 | Kobaz | use something else |
| 18:40.25 | Kobaz | number_sipfriend@regserver |
| 18:40.29 | ZcCOP | right |
| 18:40.34 | ZcCOP | and CUT based on my delimter |
| 18:40.37 | ZcCOP | kk |
| 18:40.57 | [TK]D-Fender | nullable_type: pastebin your attempt |
| 18:41.13 | Kobaz | i haven't played with that stuff too much, but that should work.. maybe |
| 18:42.05 | nullable_type | D-Fender >> It just says No Such command when i typed "show channels" in CLI |
| 18:42.16 | [TK]D-Fender | nullable_type: "core show channels concise" |
| 18:42.26 | [TK]D-Fender | nullable_type: You seem out of the loop by 2 major versions |
| 18:42.58 | nullable_type | D-Fenderr >> Lol ya that works |
| 18:43.03 | *** join/#asterisk ReDNeQ (n=ReDNeQ@70.114.226.70) |
| 18:43.48 | nullable_type | D-Fender >> To disconnect a call in progress, Is there anyway i can just hangup by giving this format for channel SIP/phoneNumber@gateway |
| 18:44.15 | nullable_type | I am trying to do via AMI from a remote machine |
| 18:44.29 | [TK]D-Fender | nullable_type: "soft hangup [channel]" |
| 18:45.09 | ZcCOP | ok so now I need some syntax help on how to parse and use sipfriendacount_5555551212, and have it call SIP/5555551212@sipfriendaccount |
| 18:45.16 | nullable_type | but the channel is usually like SIP/gateway-7digitID, It is hard to get that chanell name unless i do a show channel |
| 18:45.34 | wtca | Can voice mail greetings be stored in a DB ? |
| 18:46.08 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 18:46.44 | Kobaz | ZcCOP: use cut |
| 18:47.30 | ZcCOP | got that, looks ok based on the doc im reading.. gotta setup something to take any extension starting with a letter, to be handled.. |
| 18:47.44 | Kobaz | i thought you were gonna do exten_peer |
| 18:47.50 | Kobaz | makes matching easer |
| 18:48.16 | ZcCOP | problem is, for calls being sent the other direction, my syntax is to pass _X. back to the PRI server |
| 18:48.45 | Kobaz | set(num=${CUT(EXTEN,_,1)}) |
| 18:48.46 | Kobaz | etc |
| 18:49.09 | ZcCOP | i suppose I can change that to be more specific to only allow 10 digit to be passed back to PRI server |
| 18:49.28 | Kobaz | you should have seperate contexts for incomming and outgoing |
| 18:49.49 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 18:49.51 | *** join/#asterisk cesar_CR (n=cesar@201.201.176.2) |
| 18:49.55 | lowtek | Anyone sucessfully configured a LIP-6812 |
| 18:50.06 | lowtek | .. with asterisk (Sorry, my cut and paste had a c/r in it) |
| 18:50.14 | ZcCOP | right, but if my sippeers are setup in realtime, how does a call know to use a specific peer, which has a specific inbound/outbound context? |
| 18:50.19 | lowtek | It's an Nortel |
| 18:51.01 | Kobaz | it knows the peer since it was given one |
| 18:51.11 | ZcCOP | duh. |
| 18:51.12 | ZcCOP | sorry |
| 18:51.14 | Kobaz | you cut it out of the exten that was passed into the context |
| 18:52.14 | ZcCOP | im sure im just over complicating it, which is why im running in circles.. |
| 18:52.19 | Kobaz | heh |
| 18:53.30 | Shaun2222 | when sombody calls in and it shows there name/number whats providing this? I switched from my incomming calls comming down voicepulse to a T1, with VP i saw there name and number, with the t1 i'm just seeing a number. |
| 18:53.57 | Kobaz | Shaun2222: it comes from your telco |
| 18:54.13 | Kobaz | Shaun2222: if you just see the number, then your telco isn't giving you the name |
| 18:54.20 | Kobaz | either you need to pay extra for it, or they don't support it |
| 18:54.32 | Kobaz | more likely than not, they don't support it |
| 18:55.18 | Shaun2222 | t1 is from timewarner |
| 18:55.26 | Kobaz | call them up |
| 18:55.51 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 18:56.40 | nullable_type | Question: How can i disconnect a call via AMI using soft hangup but just by passing a phone number. Eg: SIP/phonenumber@gateway. |
| 18:57.08 | Kobaz | you'll need to get channels and find the channel you want based on a pattern match |
| 18:57.33 | nullable_type | i c |
| 18:57.43 | nullable_type | thats pretty much the only solution? |
| 18:59.02 | Kobaz | soft hangup takes a channel |
| 18:59.11 | Kobaz | so you need to give it one |
| 19:01.08 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 19:03.09 | nullable_type | ok thank you |
| 19:07.02 | *** join/#asterisk docelmo (n=vircuser@pool-70-110-115-83.lyn.east.verizon.net) |
| 19:08.04 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:08.15 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 19:15.12 | Shaun2222 | TW is saying they are not blocking Caller ID or Name |
| 19:15.21 | Shaun2222 | they want me to check my pbx. |
| 19:15.29 | Shaun2222 | so what handles that on asterisk |
| 19:16.25 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 19:17.00 | unpaidbill | anyone know of a decent callerid application that will run in the taskbar and work with 1.6? |
| 19:17.01 | carrar | over T1/PRI? |
| 19:17.11 | carrar | (Shaun2222) |
| 19:17.19 | Shaun2222 | ya |
| 19:17.21 | carrar | You need to put a wait before you answer |
| 19:17.31 | carrar | since it comes in between 1st and 2nd ring |
| 19:17.38 | carrar | Caller id NAME that is |
| 19:17.46 | carrar | not number, that comes over the D channel |
| 19:17.52 | carrar | PRI that is |
| 19:18.12 | Shaun2222 | i dont have teh ADSI module loaded, it's not part of that right? |
| 19:18.44 | *** join/#asterisk msetim (n=setim@200.195.162.210) |
| 19:18.44 | carrar | What does the ADSI module do? |
| 19:19.00 | Qwell | carrar: ADSI |
| 19:19.06 | carrar | I know |
| 19:19.09 | carrar | just making him answer it |
| 19:19.19 | Shaun2222 | dunno... http://www.voip-info.org/wiki/view/ADSI |
| 19:19.25 | Shaun2222 | looks like it's for on screen stuff |
| 19:19.39 | Shaun2222 | just dont know if it was part of the caller name stuff. |
| 19:19.49 | iMacGyver | is there a known issue with receiving faxes through a zaptel and having them email? asterisk is segfaulting partly through the fax every time. |
| 19:19.59 | carrar | not between asterisk and the pri it isn't |
| 19:20.18 | Shaun2222 | right now these are the only modules i load... http://rafb.net/p/VHyxWW49.html |
| 19:20.23 | iMacGyver | version Asterisk 1.4.22-3 |
| 19:21.01 | Shaun2222 | carrar: i have answer and wait(1.0) as the first two in my list. |
| 19:21.14 | Shaun2222 | actually.. |
| 19:21.56 | Shaun2222 | this is what i have. |
| 19:21.57 | Shaun2222 | http://rafb.net/p/u0GQPd56.html |
| 19:22.11 | Shaun2222 | from-pstn is where the incomming calls from the t1 dump. |
| 19:24.29 | [TK]D-Fender | Shaun2222: where is your failed call? |
| 19:24.44 | Shaun2222 | [TK]D-Fender: the call didnt fail |
| 19:25.04 | Shaun2222 | [TK]D-Fender: trying to figure out why calls down my t1 are not showing the caller name, and only the phone number. |
| 19:25.05 | vlt | Hello. I'm running Asterik 1.4 from Debian lenny's package. What do I have to install to receive faxes with Asterisk? |
| 19:25.20 | [TK]D-Fender | Shaun2222: What signalling? |
| 19:25.21 | *** join/#asterisk Assid (n=assid@unaffiliated/assid) |
| 19:25.35 | Assid | _ShrikE: were you the one running flowroute? |
| 19:25.38 | vlt | I tried rxfax() but that isn't available ... |
| 19:26.39 | unpaidbill | so i guess i'm going to have to roll my own call notification :/ |
| 19:26.41 | ZcCOP | Kobaz, interesting issue.. if I have a sippeer of regserver3 and a sippeer as regserver3-in, so I can have inbound and outbound using different contexts.. since both have host=1.1.1.1 (for example), its randomly selecting which peer to use |
| 19:26.49 | unpaidbill | cry with me. |
| 19:26.50 | Shaun2222 | [TK]D-Fender: pri_cpe |
| 19:27.05 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 19:27.36 | [TK]D-Fender | Shaun2222: Dump a screwed up call showing the CID, and PRI debug |
| 19:27.40 | ZcCOP | so if a call comes from PRI server to server c, it selects one peer or the other, but I cant seem to control which one.. |
| 19:27.48 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 19:27.55 | *** join/#asterisk bmoraca (n=chatzill@209.60.253.58) |
| 19:31.22 | tzafrir_laptop | vlt, you could use iaxmodem and hylafax |
| 19:31.36 | tamiel | tzafrir_laptop: About my system.conf in dahdi for two cards, I have found this old post on digium bugtracker : http://bugs.digium.com/view.php?id=2227 |
| 19:31.42 | *** join/#asterisk nicoAMG (i=asgalt@201.203.96.42) |
| 19:31.45 | tzafrir_laptop | or build the app_fax package I failed to push into Lenny |
| 19:32.10 | *** join/#asterisk mnicholson (n=mnichols@nat/digium/x-13b2f707146ba0a0) |
| 19:32.22 | *** part/#asterisk CCFL_Man2 (i=bacb35fc@pool-71-241-122-24.scr.east.verizon.net) |
| 19:33.05 | tzafrir_laptop | tamiel, in the timing column replace any number > 3 with 3 |
| 19:34.20 | tamiel | tzafrir_laptop: ok thanks :) |
| 19:44.32 | *** join/#asterisk stevetotaro (n=Steve@pool-72-72-143-197.hrbgpa.dsl-w.verizon.net) |
| 19:45.34 | *** join/#asterisk neurosys (n=vinix@sheltercorp.net) |
| 19:46.44 | j_kroon | hi all, busy fooling with cdr_mysql. I note that I can get cli and src as well as dst, is there something where I can get the name part of the cli only? |
| 19:48.06 | j_kroon | ZcCOP, not quite so randomly i believe. |
| 19:49.07 | j_kroon | ZcCOP, you should be creating 1 user and 2 peers if i'm not mistaken, which should fix your "randomness" |
| 19:50.13 | *** join/#asterisk joako (n=joako@opensuse/member/joak0) |
| 19:50.43 | joako | Can someone recommend a VoIP provider that has E911? |
| 19:51.22 | ZcCOP | yeah, i think ive got that sorted..i was making 3 equal peers (only difference was the name) for my 3 boxes.. but then it hit me.. theres no reason a call would go from server b to c, or c to b, so i ended up just making my PRI server context "outdial" and left server B and C as incoming |
| 19:51.28 | ZcCOP | that seemed to untangle the mess i made |
| 19:51.39 | ZcCOP | joako, id recommend vitelity |
| 19:52.14 | ZcCOP | they peer with XO, so their rates are very nice, and no questionable billing practices :) |
| 19:52.45 | *** join/#asterisk russellb_ (n=russell@asterisk/digium-open-source-team-lead/russellb) |
| 19:52.45 | *** mode/#asterisk [+o russellb_] by ChanServ |
| 19:59.02 | vlt | tzafrir_laptop: Ok, thank you. |
| 20:04.53 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 20:06.27 | ZcCOP | im confused.. im using Realtime() in my dialplan, but when execd, it says:The RealTime application has been deprecated in favor of the REALTIME dialplan function. |
| 20:06.38 | ZcCOP | other than CASE, i dont see the difference.. heh |
| 20:08.08 | RichardLynch | I believe case may be significant in Application names for *... |
| 20:08.26 | Qwell | ZcCOP: application != function |
| 20:09.19 | ZcCOP | oh word. |
| 20:10.08 | ZcCOP | thanks for pointing that seemingly obvious detail, ill go fix heeh |
| 20:11.07 | *** join/#asterisk docelm0 (n=chatzill@206-248-239-194.unassigned.ntelos.net) |
| 20:11.30 | docelm0 | Anyone in here familiar with the Polycom IP601 and Sidecars? More specifically how to setup the side cars from a TFTP server? |
| 20:13.12 | *** join/#asterisk intralanman (n=intralan@va-71-0-86-105.dyn.embarqhsd.net) |
| 20:14.25 | [TK]D-Fender | docelm0: You setup the phone directory as specified in the admin guide and it spills over is SD order |
| 20:14.31 | ZcCOP | docelm0: treat it like one o f the 6 line keys |
| 20:14.54 | docelm0 | Do the directory.xml file isnt where to set it up then? |
| 20:14.56 | ZcCOP | editing the mac-directory.xml file |
| 20:15.28 | docelm0 | ZcCOP: I added 40 some in there but they are not showing up on the side car.. I can see them in the directory of the phone tho |
| 20:15.28 | [TK]D-Fender | docelm0: yes |
| 20:15.34 | ZcCOP | yes, they are setup there, if you want to specifcly set the sidecar up, you start the index at 5 (or is it 6?) |
| 20:15.44 | [TK]D-Fender | docelmmay sure to set the INDEX for them in the order you want them to appear |
| 20:16.04 | docelm0 | I dont remember seeing a index directive |
| 20:16.06 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 20:16.40 | docelm0 | ZcCOP: I would imaging 7 being the 601 has 6 line keys on the phone |
| 20:17.11 | [TK]D-Fender | docelm0: No, the # doesn't really matter so long as there is no identical #. being sequential doesn't matter either |
| 20:17.16 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 20:17.21 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 20:18.26 | *** join/#asterisk ingenius (n=alektro@69.90.72.173) |
| 20:19.05 | docelm0 | then it should work then. in the line config I have setup 6 line appearences for the the extension. Then setup 0000000000-directory.xml to setup the contact directory. However when the 601 is rebooted it doesnt show anything on the side cars.. |
| 20:19.09 | docelm0 | confusing.. |
| 20:19.29 | docelm0 | Are there any other settings say in sip.cfg or something that need to be set? The admin guide is giving me a headache |
| 20:19.41 | [TK]D-Fender | docelm0: THAT is your mistake. |
| 20:20.14 | [TK]D-Fender | docelm0: The 000000 file is loaded only ONCE by the phone the first time it initializes, and then copies it over to the MAC-phone.xml version |
| 20:20.37 | docelm0 | ahh.. so I need to make the MAC-directory.xml |
| 20:20.39 | docelm0 | ok |
| 20:20.42 | docelm0 | will do that.. thanks TK |
| 20:20.56 | *** join/#asterisk rustyspoon (n=chatzill@adsl-63-195-5-177.dsl.chic01.pacbell.net) |
| 20:24.30 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 20:24.31 | *** join/#asterisk moy (n=chatzill@74.12.124.89) |
| 20:24.53 | rustyspoon | I have a server i inherited, Suddenly when phones are factory reset they no longer get dial tone. I can see them hit the server for dhcp but the phones never become active. Being a complete asterisk noob could some one possibly point me in the right direction? |
| 20:25.10 | Qwell | why would you factory reset the phones? |
| 20:25.14 | *** join/#asterisk [acer]lanman (n=intralan@va-71-0-86-105.dyn.embarqhsd.net) |
| 20:25.29 | Qwell | that makes them lose all of their configuration... meaning they have no clue how to access your server |
| 20:25.45 | carrar | heh |
| 20:25.56 | carrar | rusty, what model of phones |
| 20:26.07 | rustyspoon | Linksys SPA941 |
| 20:26.18 | rustyspoon | previously factory resets where not an issue. |
| 20:26.27 | rustyspoon | they would hit tftp pick up config and was not an issue. |
| 20:26.37 | carrar | so those come with a ROM version that it will default too |
| 20:26.56 | hardwire | rustyspoon: is dhcp set up to send tftp options to dhcp clients? |
| 20:27.52 | carrar | probably just a broken dhcp server |
| 20:27.58 | rustyspoon | Couldent tell you how its setup. I havent made any changes myself.. Where would I check to see if thats the case. |
| 20:28.04 | rustyspoon | Phone gets an IP. |
| 20:28.11 | rustyspoon | so DHCP appears to be fine |
| 20:28.20 | rustyspoon | I can ping the phone from the server. |
| 20:28.23 | ZcCOP | im having problems with REALTIME function, this keeps returning no results: Set(row=${REALTIME(sippeers,name,customer)}) |
| 20:28.47 | ZcCOP | but I know "customer" in the name feild exists in the database |
| 20:30.21 | *** join/#asterisk Micho123 (n=mcho123@77.42.151.207) |
| 20:32.03 | Micho123 | Hi all, I need to create a dialplan with extension value an array...Like 2894555[0-3],1,Do something...When dialing 28945550 I got not found...ANy cluee? |
| 20:32.45 | kaldemar | Micho123: put _ in front of the extension number |
| 20:33.12 | [TK]D-Fender | Micho123: show us your exact full dialplan context and the exact CLI output of the failed call with SIP (or whatever) debug . |
| 20:33.18 | ZcCOP | yes, like kaldemar said the "_" means match a pattern |
| 20:33.48 | Micho123 | kaldemar, Working...Thanks |
| 20:33.52 | *** join/#asterisk umpc (n=Justin@unaffiliated/umpc) |
| 20:33.56 | Micho123 | [TK]D-Fender, working fine...thanks |
| 20:34.18 | [TK]D-Fender | I did suspect that... I jsut like seeing the full evidence |
| 20:34.19 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu2.dynamic.dsl.tele.dk) |
| 20:34.31 | *** join/#asterisk ManxPowerAsteria (n=Administ@router.asteriasgi.com) |
| 20:34.54 | *** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903) |
| 20:35.30 | ZcCOP | anyone have any tips on how to debug why the REALTIME func isnt returning a result? |
| 20:38.28 | *** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com) |
| 20:38.43 | rwaite | okay, i know i have asked about this before but i am gonna do it again. |
| 20:38.49 | rustyspoon | I should mention all the other phones reboot and get configs fine. |
| 20:39.02 | rob0 | rwaite: NO! I will not marry you. |
| 20:39.14 | *** join/#asterisk juanjoc (n=juanjoc@200.69.219.113) |
| 20:40.14 | rwaite | when making an outbound call through a sip "trunk", the phone seems to answer late on the caller's end. example: if the called party sends us to voicemail, we will either not hear the message at all or hear it start late. sometimes people will pick up and hear silence and we will connect a few/ten or so seconds later |
| 20:40.41 | rwaite | is this something that can be caused by incorrectly configuring the dialplan? is it a sip problem? is it common or am i just cursed? |
| 20:42.43 | ZcCOP | argh.. REALTIME function is a joke apparently.. cant get it do function |
| 20:42.55 | ZcCOP | looks like ill need to use the deprecated application heh |
| 20:44.01 | ManxPowerAsteria | rwaite: try canreinvite=no |
| 20:45.10 | rwaite | It's set already, for the phone, and for the sip provider |
| 20:45.16 | ManxPowerAsteria | rwaite: not a dialplan issue. I've never heard of more than 1 second or so of no-audio at the start of the call, so this may not be the issue. |
| 20:45.36 | rwaite | Also, and this may help, this seemed to happen with another provider, over IAX |
| 20:46.06 | rwaite | But I'm not totally convinced this is exactly the same issue |
| 20:46.06 | rwaite | I may be wrong though |
| 20:46.19 | ManxPowerAsteria | rwaite: sounds like you might have to do some sip debugging to see if there are any delays in the signaling packets. |
| 20:46.44 | *** join/#asterisk bbryant (n=brett@c-68-59-20-153.hsd1.sc.comcast.net) |
| 20:46.52 | rwaite | When I call my cell phone, and send it to voicemail, the call picks up and I hear silence - the voicemail is already recording. It's real weird. |
| 20:47.11 | rwaite | Hmm. What could cause delays in the signaling packets? Network congestion? |
| 20:48.44 | rwaite | Any idea if progressinband could have an effect? |
| 20:48.58 | rwaite | Oh wait, I think I asked that before and it doesn't mean what I think it means |
| 20:51.38 | *** join/#asterisk AndiM (n=andi@p54A3ACFC.dip0.t-ipconnect.de) |
| 20:51.41 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 20:52.30 | AndiM | hi guys! does anyone know when doing a callback (callthrough) how to make Asterisk wait until the callback is fully established to the target device before executing further extens in that context? |
| 20:52.47 | nullable_type | in extensions.conf can i have like exten => s,n,EXECUTE_A_LINUX_COMMAND? |
| 20:53.01 | [TK]D-Fender | nullable_type: System() |
| 20:53.04 | AndiM | when doing GSM calls call setup takes ages and this is too long since several commands will have been executed then. |
| 20:53.12 | nullable_type | D-Fender >> Thanks! |
| 20:53.16 | AndiM | nullable_type: and load app_system.so or so. |
| 20:53.44 | [TK]D-Fender | AndiM: Either * thinks the call is fully established, or it doesn't. Blame goes to your interface |
| 20:54.19 | AndiM | [TK]D-Fender: oook. Somehow suspected so. Any hints where to look at this? |
| 20:54.22 | nullable_type | I was thinking of doing of doing something just when a call end using System command |
| 20:54.29 | nullable_type | does that sound terrible? |
| 20:54.32 | [TK]D-Fender | AndiM: Your interface <- |
| 20:55.14 | [TK]D-Fender | nullable_type: Depends what you want to do. If you were to say "electrocute a kitten", I'd have to say "yes" |
| 20:55.25 | AndiM | <[TK]D-Fender: that callback call goes out through a Local/ which goes through an IAX/ in another extension and then goes through another SIP/ on the peer side ;) Anyway, I'll analyse this. |
| 20:55.33 | jplank | hahahah |
| 20:55.38 | AndiM | s/extension/context/ |
| 20:55.43 | nullable_type | D-Fenderer > lol if i were to say like a simple wget |
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| 21:00.15 | [TK]D-Fender | nullable_type: Complexity isn the question. |
| 21:00.29 | [TK]D-Fender | nullable_type: more than fine |
| 21:01.57 | nullable_type | alright, thanks |
| 21:03.58 | *** join/#asterisk bkw_ (n=brian@freeswitch/developer/bkw) |
| 21:04.11 | rwaite | How would I determine if the signaling is delayed by looking at sip debug? |
| 21:04.32 | [TK]D-Fender | rwaite: yes :) |
| 21:05.05 | rwaite | I'm gonna have to read the sip documents aren't I |
| 21:07.10 | Qwell | 1 |
| 21:07.12 | Qwell | opops |
| 21:08.01 | *** join/#asterisk AndyML (n=quassel@pool-98-111-140-229.phlapa.fios.verizon.net) |
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| 21:09.25 | ZcCOP | baring the fact that the REALTIME function didnt do anything, im using the deprecated Realtime application, its working now.. |
| 21:09.47 | ZcCOP | now the fun task of building thousands of numbers in a realtime database (each with 3 steps).. |
| 21:09.52 | rwaite | chan_sip.c: Strict routing enforced for session << what is this? |
| 21:15.24 | *** part/#asterisk ManxPowerAsteria (n=Administ@router.asteriasgi.com) |
| 21:16.13 | AndiM | [TK]D-Fender: GOT IT! the destination (peer's) dialout context had a Background() which didn't have 'n' ("don't answer the channel"), thus my channel handling got answered there and the channel was ready at the originating peers side and thus exten processing started too early. |
| 21:18.30 | AndiM | now I know why it happened - I just don't know whether my _entire_ organisation has proper structural setup given those Background()s at the wrong side of the routing... |
| 21:19.43 | rwaite | so my theory is this - when the call goes out, the provider sends my server a message saying "calling this guy" and asterisk tells my phone to start ringing, then when the call connects, the provider and my server start talking all RTP-like, and at the same time sends a message to stop ringing - if there is any delay in the message to tell it to stop ringing, my end may keep hearing ringing until it gets that packet |
| 21:19.47 | rwaite | does that sound correct? |
| 21:20.19 | jplank | in the very loosest of turns, that sounds like SIP |
| 21:20.48 | jplank | s/turns/terms |
| 21:21.11 | AndiM | rwaite: if there's a severe disconnect in your ringing handling (keeps ringing and ringing and ringing), then it is said that this may be caused by NAT issues... |
| 21:21.42 | rwaite | well that sucks, can't my asterisk server know that it should probably stop ringing as soon as the rtp connection is established? |
| 21:22.16 | rwaite | heh, well if its nat issues its probably the fact that nat exists at all. |
| 21:22.18 | AndiM | but if it's just that there's a semi-long but already annoying ringing delay, then of course it probably isn't NAT issue. |
| 21:22.36 | rwaite | i dont see the difference? |
| 21:23.03 | rwaite | its intermittent too, sometimes there is no delay, other times, bad delay. thats what makes me think it might be congestion. but the network seems fine. ugh. |
| 21:23.48 | AndiM | rwaite: network load is negligible? Otherwise try beefing up your QoS. |
| 21:25.02 | AndiM | rwaite: or simply try a "watch -n 2 "ifconfig <outgoing_iface>"" to see what kind of potentially lag-prone traffic you have... |
| 21:25.06 | rwaite | i might try simplifying it, and just prefering all traffic from the asterisk box |
| 21:25.49 | *** join/#asterisk nima0102 (n=nima@91.98.206.161) |
| 21:27.29 | aaroneous | hmm.. anyone here know about BRIs and if they are suitable for companies that need the features of a PRI but only use around 8-10 channels? |
| 21:27.46 | Assid | dd-wrt needs to fix its qos |
| 21:30.40 | AndiM | Assid: hmm why? |
| 21:31.07 | Assid | major issue in the way it handles it |
| 21:31.08 | *** join/#asterisk [intra]lanman (n=intralan@freeswitch/developer/intralanman) |
| 21:31.21 | Assid | pre 24sp1 it used to work fine |
| 21:31.30 | Assid | after that.. messed up |
| 21:31.31 | AndiM | Assid: you mean config-wise, not quality-wise, right? |
| 21:31.37 | Assid | quality wise |
| 21:31.42 | Assid | config is the sam |
| 21:31.45 | Assid | same |
| 21:32.05 | AndiM | Assid: using HTB etc. QoS on Asterisk OpenWrt white-russian here. |
| 21:32.20 | Assid | hrmm im thinking of moving to openwrt |
| 21:32.28 | Assid | dunno if i will end up on it |
| 21:32.31 | AndiM | Assid: and I'm not entirely happy either ;) |
| 21:32.39 | Assid | try tomato |
| 21:32.45 | rwaite | when i call, i get a message on the console about packet2packet bridging |
| 21:32.47 | AndiM | Assid: (VoIP QoS seems _hard_ to get right) |
| 21:32.51 | rwaite | could this have anything to do? |
| 21:33.04 | Assid | AndiM: dd-wrt's QoS worked brilliant before |
| 21:33.16 | rwaite | Tomato seems to work well for me QoS-wise |
| 21:33.35 | AndiM | g'damn, too many router distro users here ;)) |
| 21:33.35 | Assid | lets just say i used tro download and use voip and have another laptop on the same time and voip worked brilliant |
| 21:34.03 | Assid | all this on a 128kbit line |
| 21:34.06 | Assid | hehe |
| 21:34.15 | AndiM | Assid: "download" or "massive, router-crashing download"? I'm coping with the latter one sometimes... |
| 21:34.24 | AndiM | (ok, almost crashing ;) |
| 21:34.33 | Assid | p2p/torrent |
| 21:34.58 | Assid | + surfing and such + laptop doing general surfing/emailing/messengers |
| 21:35.13 | AndiM | used to have a full dormitory floor with 20 rooms all connected to one measly P75/16MB w/ 1Mb DSL and still _tolerable_ traffic lag... |
| 21:35.32 | Assid | i should try openwrt |
| 21:35.35 | Assid | maybe this weekend |
| 21:35.45 | Assid | if i cna get someone to help me wit it |
| 21:36.25 | AndiM | and I should try openwrt/2.6 (my 2.4 installation is somewhat bad w/ Asterisk and I'm confident with 2.6 it would be _much_ better) |
| 21:36.52 | AndiM | (that 2.4.x kernel has _awful_ latencies on I/O) |
| 21:36.54 | Assid | 2.6 ? you mean asterisk version or wrt version |
| 21:37.46 | AndiM | but I really fear that massive upgrade from a fully tweaked rotten white-russian install... |
| 21:41.42 | rwaite | Well, it seems to work better if I use ulaw internally and g729 to connect to the provider (before I was using g729 as well) this prevents it from doing packet2packet bridging (which I think passes the audio between the phone and the provider's server more or less without touching it) |
| 21:41.59 | rwaite | I would expect this to result in /worse/ latency/delay, but hey, it seems to be working |
| 21:42.26 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 21:43.54 | AndiM | rwaite: maybe some jitterbuffer settings influence your issues? |
| 21:44.16 | rwaite | I've not done anything with the jitterbuffer |
| 21:46.23 | AndiM | rwaite: ...which could be a mistake precisely ;)) |
| 21:46.56 | rwaite | I'm looking at it, I'm looking at a lot of stuff heh |
| 21:47.15 | AndiM | rwaite: disclaimer: I'm an * greenhorne (however two peer sides plus IAX plus callback setup now, so, well, semi-greenhorne maybe) |
| 21:49.28 | *** join/#asterisk sasargen (n=chatzill@70-4-5-90.pools.spcsdns.net) |
| 21:49.32 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 21:50.05 | *** join/#asterisk timeshell_atwork (n=chatzill@gw.lusi.on.ca) |
| 21:52.10 | rwaite | heh, it's cool. |
| 21:53.46 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 21:54.49 | rwaite | i've been working w/* since feb 2008 and i'm still extremely inexperienced |
| 21:57.10 | *** join/#asterisk A500mg (n=x@ACaen-156-1-34-222.w90-51.abo.wanadoo.fr) |
| 21:57.14 | A500mg | hello :) |
| 21:58.10 | A500mg | i've a little question |
| 21:58.59 | A500mg | i've a material who support only zaptel, no dahdi |
| 21:59.33 | A500mg | but i need to use asterisk 1.6 ( not compatible with zaptel) |
| 21:59.57 | [TK]D-Fender | A500mg: which> |
| 22:00.23 | rob0 | DAHDI was merely a name change of the *same* driver. |
| 22:00.34 | A500mg | mmh |
| 22:00.52 | rob0 | Well, a slight restructuring, split into driver and software, but still, same thing. |
| 22:01.21 | A500mg | my material is an usb dongle who just provide a timer (i don't want dahdidummy) |
| 22:01.51 | A500mg | ok so i can rename all "zaptel" words by "dahdi" ..... |
| 22:02.32 | AndiM | whoa, timer provided via USB? so far all I heard was latency _caused_ by USB transactions... didn't think that such a thing would be done with USB, but then what do I know... |
| 22:03.03 | A500mg | yes but i've test with zaptel and it work :) |
| 22:03.21 | *** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no) |
| 22:03.28 | A500mg | i prefer use a tdm400p, but the single pci port is for the b410p |
| 22:04.48 | [TK]D-Fender | A500mg: your PCI card should provide timing |
| 22:05.43 | A500mg | yes if i use wcb4xxp driver of dahdi |
| 22:05.55 | A500mg | with misdn, no ? |
| 22:06.24 | [TK]D-Fender | A500mg: probably |
| 22:06.40 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 22:08.57 | A500mg | wcb4xxp is a little "young" for production usage |
| 22:09.08 | A500mg | i prefer misdn for the moment |
| 22:09.34 | A500mg | i will test wcb4xxp in a little time .. |
| 22:10.06 | *** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) |
| 22:10.53 | A500mg | question: with a tdm400p without fxo/fxs modules (like a "tdm00b"), it is possible to have a timer ? |
| 22:11.23 | rwaite | can asterisk send sms messages? |
| 22:13.41 | A500mg | yes rwaite |
| 22:13.59 | RichardLynch | also wonders if * can send SMS, but hasn't read enough docs to find out if it's a stoopid question... |
| 22:22.28 | *** join/#asterisk bullium (n=will@216.68.250.30) |
| 22:25.31 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 22:27.00 | nullable_type | Hey guys, In my extensions.conf if i have |
| 22:27.01 | nullable_type | exten => s,4,DIAL() |
| 22:27.01 | nullable_type | exten => s,5,DoSomething() |
| 22:27.13 | nullable_type | will DoSomething() executed after channels hang up? |
| 22:28.34 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 22:28.38 | jksM | hi guys.. are recomendations on a cheap way to convert approx. 125 analog phones to SIP in a "not too expensive" way? - I have been looking at audiocodes MP-100 series... but I have would to get a bundle of 24 port boxes... |
| 22:30.57 | Talkradio | you can probably find phones on ebay cheaper |
| 22:31.09 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 22:31.41 | nullable_type | Talkradio Did you look at my question above, I need some help with Extension.conf |
| 22:31.52 | Talkradio | i'm a newb sry |
| 22:32.02 | nullable_type | no worries |
| 22:32.08 | nullable_type | newb myself too |
| 22:33.06 | jksM | Talkradio, can't do phones, need to convert the old analog phones |
| 22:33.32 | Talkradio | only way is use ata's |
| 22:33.55 | jksM | Talkradio, yes, but can you recommend one with a lot of ports? |
| 22:34.21 | Talkradio | nope |
| 22:34.58 | Talkradio | what are you doing trying to convert old system to asterisk |
| 22:35.15 | jksM | something like that, yes |
| 22:35.51 | Talkradio | might be easier to just buy card for pc to accept analog lines |
| 22:36.19 | jksM | can you recommend any with a lot of ports? |
| 22:36.48 | Talkradio | sagnoma makes them and i'd look on www.voipsupply.com |
| 22:37.21 | jksM | yes, okay - I know the normal sangoma cards - but I cannot find any suitable for this |
| 22:38.25 | Talkradio | call them they are good to bounce things off of |
| 22:38.37 | [TK]D-Fender | jksM: AudioCodes MP-124 = recommendable |
| 22:38.40 | jksM | well, I was more looking for recommendations... i.e. stuff that people actually tried out in real life |
| 22:38.41 | Talkradio | 1-800-398-8647 |
| 22:38.53 | [TK]D-Fender | nullable_type: No. "h" Asterisk Standard Extension. |
| 22:39.01 | jksM | putting 5-6 of the 24 port sangoma cards in one server... I don't know if that's a good idea |
| 22:39.14 | [TK]D-Fender | jksM: horrible |
| 22:39.32 | *** join/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 22:39.34 | Talkradio | don't they have bigger than 24? |
| 22:39.51 | jksM | [TK]D-Fender, I had suspected that :-) (regarding AudioCodes).... but still having 6 of those boxes seems a bit complex and perhaps costly |
| 22:39.54 | *** join/#asterisk killown (n=ukendt@unaffiliated/killown) |
| 22:40.29 | [TK]D-Fender | jksM: A little slow to learn at the start but exporting configs to configure multiple units after becomes a cinch |
| 22:40.32 | jksM | [TK]D-Fender, do you know anything about the pricing of their blade system? I'm thinking it's probably a lot more expensive, and the MP-124s might be least costly thing afterall |
| 22:40.43 | jksM | Talkradio, not that I know of_ |
| 22:41.55 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net) |
| 22:42.28 | jksM | Talkradio, the 24 port solution they have is actually 2 cards... so I would have to have 12 pci cards installed.. 18 if I go with the echo canceller aswell... I doubt that will work well, unless I buy some kind of monster server with lots of pci busses |
| 22:43.46 | *** part/#asterisk dlewis (i=4579ba4a@about/security/staff/dlewis) |
| 22:44.20 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 22:45.09 | [TK]D-Fender | jksM: just forget that idea outright. MP-124 = way to go |
| 22:45.23 | Talkradio | i've only done 4 port a200 cards or T1 card.. nothing big |
| 22:45.26 | jksM | [TK]D-Fender, okay, you're not recommending the blade solution? |
| 22:45.41 | jksM | [TK]D-Fender, (I know the PCI card solution is flawed from the start) :-) |
| 22:47.16 | [TK]D-Fender | jksM: "blade"? Pardon? |
| 22:47.36 | jksM | [TK]D-Fender, the audiocodes blade system that I wrote about before? |
| 22:48.13 | [TK]D-Fender | jksM: I don't see a reference to BLADE anywhere. LINK please |
| 22:48.23 | jksM | [TK]D-Fender, http://www.audiocodes.com/products/tp-8410 |
| 22:48.50 | jksM | I was mainly trying to find something in between of the mp-124 which has 24 ports, and the blade solutions which apparently scales up to 2016 ports |
| 22:49.03 | [TK]D-Fender | jksM: Holy shit thats 42 T1/E1... NOT ANALOG |
| 22:49.27 | jksM | yep ;-) |
| 22:49.48 | [TK]D-Fender | jksM: You are insane for even looking at it. just get 5 x MP-124 and you're DONE |
| 22:50.12 | jksM | hehe, I was trying to find some kind of middle ground |
| 22:50.22 | jksM | if they had like a 75 port device or something, that would be nice |
| 22:50.23 | [TK]D-Fender | jksM: There is vertually none |
| 22:50.27 | [TK]D-Fender | virtually* |
| 22:50.53 | jksM | I need to find a cheaper place to buy them then :-) |
| 22:51.44 | jksM | hmm, or perhaps it's really not that bad...1495 USD |
| 22:52.51 | jksM | [TK]D-Fender, all the pictures of the mp-124 are of the front... what kind of connectors does it have? |
| 22:54.00 | [TK]D-Fender | jksM: Amphenol, Ethernet, + 1 FXO failover |
| 22:54.39 | jksM | amphenol is that weird connector which is then split up into a lot of others_ |
| 22:55.14 | jksM | i.e. one cable into the mp-124, and then into a 24 ports patchpanel kind of thing? |
| 22:55.22 | nullable_type | In Extensions.conf how do i get it when a call is complete after exten => s,n,Dial() |
| 22:56.25 | [TK]D-Fender | jksM: http://www.telephonyware.com/telephonyware/tw00610.html |
| 22:56.28 | *** join/#asterisk `Sean (n=Un1x@CPE000024cb557c-CM00125572fe2a.cpe.net.cable.rogers.com) |
| 22:56.41 | [TK]D-Fender | jksM: I've also tried the 1124 from them before. Also a solid choice |
| 22:56.46 | [TK]D-Fender | jksM: Much cheaper |
| 22:56.51 | jksM | [TK]D-Fender, oh - a lot cheaper, yes! |
| 22:56.56 | [TK]D-Fender | nullable_type: I' already answered you <- |
| 22:57.06 | nullable_type | D-Fender >> Sorry, I will refer back |
| 22:57.48 | `Sean | [TK]D-Fender, had a question lets say i had a P4 3.0Ghz with 1gb ram would it be enough to support lets say 10 Simulataneous G729 calls? |
| 22:57.53 | jksM | [TK]D-Fender, have to find out where to buy those around here :-) |
| 22:58.10 | A500mg | yes Sean, more than 10 |
| 22:58.12 | A500mg | more more more |
| 22:58.16 | `Sean | Cool :) |
| 22:58.41 | jksM | [TK]D-Fender, oh, I found a 48 port gateway from Vega |
| 22:59.06 | voxter | With PRI, I've set my LBO to 7.5 for the 220-330ft range. Do I need to instruct my carrier to set the same? Im getting dropped calls on the line. |
| 22:59.57 | nullable_type | D-Fender, I am still not sure how i can get to a line in Extensions.conf when a channel is hungup and call is complete (between two channels). Could you please tell me? |
| 23:00.10 | *** join/#asterisk riddlebox (n=user@mscitspubwlgw.wustl.edu) |
| 23:00.12 | jksM | [TK]D-Fender, http://www.patton.com/products/pe_products.asp?category=364&MiDAS_SessionID=1d07e865037e4b039adae7451b654d14 and this 32 port one from Patton |
| 23:02.04 | A500mg | and astribank ..? |
| 23:02.45 | A500mg | (maybe not enough port) |
| 23:04.06 | A500mg | question: how meetme app (for example) does it make to achieve the timer provide by ztdummy / dahdidummy ? does it achieve it by the intermediary of asterisk and chan_zap / chan_dahdi, or more "directly" ? |
| 23:04.54 | Qwell | ...what? |
| 23:05.17 | A500mg | example: i've Asterisk 1.6, with NO chan_zap (normal...), and zaptel with a functionnal ztdummy. Meetme application can use the timer, or not ? |
| 23:05.19 | A500mg | lol |
| 23:05.27 | A500mg | (sorry i don't speak english very well) |
| 23:05.27 | [TK]D-Fender | `Sean: Can handle a whole lot more than that |
| 23:05.32 | Qwell | 1.6 doesn't use Zaptel, so no. |
| 23:05.48 | *** join/#asterisk BadHAL (n=nn@cpe-72-179-194-139.stx.res.rr.com) |
| 23:05.48 | A500mg | i know that |
| 23:05.52 | [TK]D-Fender | jksM: I've heard some very depressing reviews of Patton |
| 23:05.55 | A500mg | it's not the question :) |
| 23:06.07 | jksM | [TK]D-Fender, looked to good to be true |
| 23:06.33 | [TK]D-Fender | jksM: AudioCodes or Mediatrix are the way to go for this. |
| 23:06.41 | [TK]D-Fender | jksM: Get busy soursing... |
| 23:06.44 | [TK]D-Fender | sourcing* |
| 23:06.53 | jksM | yeah, seems like mediatrix doesn't have a distributor or reseller here |
| 23:06.53 | A500mg | i don't know how asterisk use a timer provided by dahdi/zaptel: is it with his chan_zap / chan_dahdi ? |
| 23:07.01 | nullable_type | D-Fender >> Did you see my Question?! Thanks :) |
| 23:07.34 | A500mg | (it's a dumb question) |
| 23:09.05 | *** join/#asterisk Coolthreads (n=Coolthre@203-97-238-71.cable.telstraclear.net) |
| 23:09.33 | [TK]D-Fender | nullable_type: [17:38]<[TK]D-Fender>nullable_type: No. "h" Asterisk Standard Extension. |
| 23:09.49 | [TK]D-Fender | nullable_type: Go read the BASICS. This is a 101 grade question. |
| 23:10.00 | [TK]D-Fender | nullable_type: h = HANGUP. WIKI ti up. |
| 23:10.03 | [TK]D-Fender | ~wikis |
| 23:10.04 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 23:10.23 | [TK]D-Fender | off to martial arts, bbiab |
| 23:10.42 | Micho123 | Hi all, Doen anyone knows how to read a variable that I defined in a specific context on asterisk on a perl AGI script?The variable is defined on asterisk as envirement variable |
| 23:11.37 | A500mg | just a normal variable, or global variable ? |
| 23:12.05 | nullable_type | D-Fender >> Did you read my question? |
| 23:12.22 | nullable_type | Just kidding, I am reading the Wiki. Sir! :) |
| 23:12.26 | Micho123 | A500mg, Just need to create a new variable in context and read it from the perl script...it could be env variable or anything |
| 23:12.44 | A500mg | oh ok |
| 23:12.46 | A500mg | wait .. |
| 23:12.53 | Micho123 | Just need to pass this variable in order to be read from Perl script |
| 23:13.21 | Micho123 | A500mg, waiting...thaks |
| 23:17.30 | *** join/#asterisk italorossi (n=italoros@201.76.154.111.intranet.digi.com.br) |
| 23:17.45 | RichardLynch | Micho123: Google for "asterisk perl variables" second hit is this: http://www.voip-info.org/tiki-index.php?page=Asterisk+perl+agi |
| 23:17.51 | A500mg | my $number = $ARGV[0]; |
| 23:17.54 | A500mg | try this ... |
| 23:17.57 | RichardLynch | I don't do perl, but I suspect it has what you want... |
| 23:17.59 | A500mg | (example |
| 23:19.24 | italorossi | Getting this error with asterisk 1.4 and a2billing 1.3.4: CHANNEL STATUS : -1 = There is no channel that matches SIP/XXXXX |
| 23:19.51 | Micho123 | A500mg, My problem is not here...As you know there is a predefined variable in asterisk tlike accountcode, callerid hat we can use in perl script... |
| 23:20.08 | RichardLynch | Is it really XXXXX? Does sip.conf have XXXXX? Look for a typo. Restart asterisk. |
| 23:20.11 | Micho123 | A500mg, the problem if I need to add a new variable to these predefind variable |
| 23:20.36 | hardwire | e164.org is freaking out.. maaaaan.. |
| 23:20.40 | italorossi | RichardLynch: No :), it's SIP/9207630922-0826a6c0 |
| 23:20.47 | hardwire | it keeps calling my phone for verification.. but fails to prompt me |
| 23:20.52 | hardwire | I wonder if they are having super issues at |
| 23:20.52 | hardwire | m |
| 23:20.58 | *** join/#asterisk nullable_type (n=kumana@hq.verbx.net) |
| 23:21.12 | Micho123 | A500mg, is there any way to do that? I have created a global and envirement variables but cannot get them is perl script |
| 23:21.29 | nullable_type | Hey D-fender >> Thank you so much, 'h' is what i was looking for. Worked perfectly.... |
| 23:21.49 | *** join/#asterisk ghento (n=ghento@d75-157-192-235.bchsia.telus.net) |
| 23:22.29 | RichardLynch | You have that in sip.conf Copy / Paste the numbers/hex to be sure... |
| 23:23.15 | italorossi | RichardLynch: This channel exists, but the problem is: a2billing can't get channel status with phpagi.. |
| 23:23.59 | *** join/#asterisk bijit (n=chatzill@216.25.163.115) |
| 23:24.09 | RichardLynch | [shrug] Could be. The message I see says there is no such channel... Try stripping off the -082a6c0 bit. That looks like a random token to me. |
| 23:24.35 | bijit | can asterisk query a number from a db and then send it has the caller id? |
| 23:25.18 | bkw_ | looks around |
| 23:25.34 | RichardLynch | bijit: Yes. |
| 23:26.02 | bijit | really ? what should I be reading to accomplish this? |
| 23:26.26 | italorossi | RichardLynch: no... The channel must have this "random" digits.. without them is a simple sip peer.. |
| 23:26.47 | RichardLynch | It's not even tricky, really. Read the asterisk book, especially the OriginateCall bit, and snag PHP API (or Perl if you like) to glue DB in. |
| 23:26.50 | A500mg | mmmh Micho123 i don't exactly understand the question, i read english like a french :) |
| 23:27.04 | Micho123 | A500mg, ok |
| 23:27.15 | bijit | RichardLynch: thanks |
| 23:27.20 | A500mg | i go to bed ! |
| 23:27.21 | A500mg | ++ |
| 23:27.29 | *** part/#asterisk RichardLynch (n=Richard_@198.199.238.13) |
| 23:28.55 | Damin | OK.. |
| 23:29.07 | Damin | Who wants to excercise their dialplan kung fu? |
| 23:29.41 | Micho123 | Does anyone knows how to add a new variable to the predefined variables sent by asterisk to AGI script? |
| 23:29.49 | Damin | I need an statement that does the following (if callerid="" then set callerid(nam)=Anonymous) |
| 23:32.09 | hardwire | lots of examples online |
| 23:32.12 | hardwire | voip-info.org |
| 23:32.13 | hardwire | etc.. |
| 23:32.22 | *** part/#asterisk italorossi (n=italoros@201.76.154.111.intranet.digi.com.br) |
| 23:32.32 | Qwell | Damin: name or num? |
| 23:32.35 | hardwire | Damin: check out the gotoif function on voip-info.org (use the search) |
| 23:32.53 | Qwell | nah, ExecIf is your friend there :D |
| 23:33.07 | hardwire | I forgot about execif |
| 23:34.10 | Qwell | ExecIf(${ISNULL(${CALLERID(num)})},Set(${CALLERID(name)}=Anonymous)) |
| 23:34.26 | Qwell | something like that ;p |
| 23:35.45 | murdock_ut | Can you specify the RFC2833 Payload type in *? |
| 23:37.45 | Damin | Qwell: I'm trying something like this.. ExecIf(["${CALLERID(name)}" = ""]|set(CALLERID(name)="Anonymous")|1 |
| 23:38.11 | Qwell | what's the random 1 for? |
| 23:38.15 | Damin | But it's telling me app_exec.c:186 execif_exec: Could not find application! (set(CALLERID(name)="Anonymous") |
| 23:38.18 | Qwell | and you're missing a $ for your $[] |
| 23:38.51 | Qwell | oh, and I butchered the execif syntax. |
| 23:38.56 | Qwell | it's ?, not | or , |
| 23:39.18 | *** join/#asterisk egypcio (n=egypcio@unaffiliated/egypcio) |
| 23:39.20 | Qwell | ExecIf($["${CALLERID(name)}" = ""]?Set(CALLERID(name)="Anonymous")) |
| 23:39.23 | Qwell | something like that |
| 23:40.41 | Damin | <PROTECTED> |
| 23:40.43 | Damin | [2009-03-03 18:40:25] WARNING[22419]: app_exec.c:186 execif_exec: Could not find application! (Set(=Anonymous)) |
| 23:41.01 | Damin | That's using the following: exten => _2169203079,n,ExecIf($["${CALLERID(name)}" = ""]|Set(CALLERID(name)="Anonymous")) |
| 23:41.10 | Qwell | replace the | with a ? |
| 23:41.23 | Damin | Qwell: 1.4? I thought that was only for 1.6? |
| 23:41.30 | Qwell | oh, umm |
| 23:42.24 | Damin | <PROTECTED> |
| 23:42.25 | Damin | [2009-03-03 18:42:09] ERROR[22479]: app_exec.c:191 execif_exec: Invalid Syntax. |
| 23:42.37 | Qwell | yeah, guess it's ? in 1.6 only |
| 23:43.06 | Qwell | this makes no sense |
| 23:43.25 | *** part/#asterisk AndiM (n=andi@p54A3ACFC.dip0.t-ipconnect.de) |
| 23:43.41 | Damin | <PROTECTED> |
| 23:43.43 | Damin | [2009-03-03 18:42:40] WARNING[22481]: app_exec.c:186 execif_exec: Could not find application! (Set(CALLERID(name)="Anonymous")) |
| 23:43.45 | Qwell | Apparently ExecIf is completely non-functional in 1.4 |
| 23:45.18 | Damin | Qwell: Great....... |
| 23:45.54 | *** join/#asterisk path_ (n=ioctl@pc-15-190-86-200.cm.vtr.net) |
| 23:46.11 | Damin | Qwell: Hell.. this doesn't even work.. |
| 23:46.12 | Damin | exten => _2169203079,n,ExecIf($["" = ""]|Noop() |
| 23:46.21 | Qwell | yeah.. |
| 23:46.26 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 23:46.32 | Damin | [2009-03-03 18:45:49] WARNING[22571]: app_exec.c:186 execif_exec: Could not find application! (Noop() |
| 23:46.34 | Damin | <PROTECTED> |
| 23:46.41 | Damin | Qwell: FIX IT! :) |
| 23:47.26 | bmoraca | make some goto spaghetti code! |
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| 23:49.14 | Qwell | oh |
| 23:49.34 | Qwell | duh |
| 23:50.00 | Qwell | ExecIf(1|NoOp|foo) |
| 23:50.05 | Qwell | instead of... |
| 23:50.08 | Qwell | ExecIf(1|NoOp(foo)) |
| 23:50.30 | Qwell | so like |
| 23:50.44 | Qwell | ExecIf($["${CALLERID(name)}" = ""]|Set|CALLERID(name)="Anonymous") |
| 23:50.45 | Qwell | that |
| 23:51.05 | *** part/#asterisk Khratos (n=khratos@190.166.103.111) |
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| 23:57.08 | bmoraca | i see what you did there |
| 23:57.13 | bmoraca | makes sense looking at it |