IRC log for #asterisk on 20090916

00:01.06*** part/#asterisk nny (n=scott@64.203.237.47)
00:01.36*** join/#asterisk lordmortis (n=lordmort@203-59-207-20.dyn.iinet.net.au)
00:03.27phixsoho gear? what that?
00:03.35phixsmall office home office?
00:04.15Chainsawphix: Read: Silly consumer-grade stuff
00:04.42Mangobmoraca: What soho gear specifically?
00:06.35phixah ok
00:06.51MangoEh, some soho gear isn't bad.
00:06.56MangoPAP2T :)
00:07.02phixhmmmm, do I want a Linksys SPA 942 or not :\
00:07.08bmoracaMango: all SOHO gear is terrible.
00:07.15bmoracaphix: no
00:07.17Mangophix: You can do better, likely.
00:07.21voipmonkspa 942's arent horrible
00:07.26voipmonkbut the polycoms sound better
00:07.27voipmonk:)
00:07.32MangoYeah.
00:08.15MangoThe sound quality is pretty good.  Far better than a POTS phone of course.  But, there are other phones (Polycom) in the same price range with more features and yet others (Aastra) in a slightly lower price range, also with more features.
00:08.30Mangobmoraca: Hehe.  What in particular were you working with?
00:09.21phixvoipmonk: which polycom though
00:09.29voipmonkhow many lines do you need?
00:09.48phixvoipmonk: I can get some polycom 300 somethings, but apparntly they are worste than a SPA 942
00:10.02phixvoipmonk: 2 - 4 inclusive
00:10.10voipmonk"apparently" ?
00:10.26phixvoipmonk: yes, according to [TK]D-Fender :)
00:12.09voipmonkits up to you and your budget :)
00:12.45*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
00:14.15*** join/#asterisk grabes222 (n=grabes@72.20.207.237)
00:14.19phixvoipmonk: hmmm well what additional features would a same price ranged polycom have over a 942?
00:14.54voipmonki find the sound is different
00:15.02voipmonkbut I personally own a spa 942 on my desk at home
00:15.11voipmonki dont use it anymore tho :)
00:15.14voipmonki use an iaxy
00:15.17grabes222Anyone having any issues with cdr_adaptive_odbc not logging?  Its registered, the ODBC connection is valid no errors are happening.. *1.6.1.6
00:15.20voipmonkand a cordless
00:15.21bmoracaMango:  it was a crappy 8 port VPN "router" from NetGear.  but that's besides the point.  it's all garbage
00:15.40phixvoipmonk: iaxy?
00:15.51voipmonkyes... :)
00:15.53voipmonkan iaxy
00:15.59phixwhich is/'?
00:16.07phixand what type of cordless?
00:16.33bmoracadoes Pickup() work fairly well now?  my last experiences with it weren't all that great...
00:19.10*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
00:21.18*** join/#asterisk psilikon (n=psilikon@140-1.35-65.tampabay.res.rr.com)
00:24.58riddleboxbmoraca, I've never had issues with it
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00:28.50*** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk)
00:36.46phixhmmmm, Power over Ethernet
00:37.07phixany issues linking a POE switch to another POE switch?
00:40.39Nuggetdon't cross the streams
00:41.38coppicethat rule is only for pissing contests
00:41.40riddleboxlol
00:41.50riddleboxand the proton packs
00:42.34phixhaha
00:42.58phixThat didn't actually answer my question correctly :)
00:43.07phixare you saying I should or shouldn't do that?
00:48.31*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
00:48.31*** mode/#asterisk [+o leifmadsen] by ChanServ
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00:50.19drmessanoWhy worry?  Each of us is carrying an unlicensed nuclear accelerator on his back...
00:50.22drmessanoSwitch me on
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00:55.19*** mode/#asterisk [+o Deeewayne] by ChanServ
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01:25.17m477auAnyone around able to help me with invoking a agi script?
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02:05.06EnrGEhi folks
02:05.19MangoHi.
02:05.21b1u3m3thhello
02:05.28EnrGE:)
02:07.21EnrGEintro: enrge location: fj since m comparatively new to irc
02:15.52drmessanoOh a newbie
02:16.26drmessanosharpens his knives and polishes his ~ key
02:20.45*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
02:25.30EnrGElol drmessano
02:31.42carrarw00t
02:32.32*** join/#asterisk lordmortis (n=lordmort@203-8-160-250.secure.com.au)
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02:41.22dandate2comcast cable business services or phillipine E1 line
02:42.30drmessanoThey can't extradite you if you move to ecuador
02:43.08dandate2lol
02:43.23dandate2no i am in a small village of the phillipines ain't no extradition here
02:43.38dandate2noone even checks visas heh
02:44.17carrarI'm in a small village also
02:44.28carrarvillage of a few million!
02:44.36carrarwell thousands
02:45.17dandate2my agents still complain of line issues at times, and random disconnects in the middle of the call. i'd like to just say cuz they hung up on your ass but wondering if a $1000/mo phillipine leased wire will do better than american digital
02:45.56carrarIsn't SIP service from America to the phillipines Illegal?
02:46.17carrar(in phillipine)
02:46.33dandate2yes it is
02:46.39dandate2but i am a foreign national
02:46.42dandate2so noone here cares
02:46.43carrarhaha
02:46.51carraroh
02:46.54carrarthen it's ok
02:46.57dandate2yes
02:46.59carrarhaha
02:47.58carrarNot sure we can support your illegal activities here
02:48.06carrarMr. Criminal Man
02:48.17dandate2haha
02:48.19dandate2whatever
02:48.23dandate2screw smart
02:48.24dandate2and globe
02:48.28dandate2those bastards
02:48.43carrarFather Internet would be sad
02:48.46dandate2lol
02:48.55carrarFather Internet: http://pics.osburn.com/photo/42882/original
02:49.09dandate2the telco companies herein the phillipines offer ridiculously bad service but have made the government outlaw all competition to them
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02:49.33dandate2the phillipine gov and telco companies would like to see your people living in tiki huts eating bananas and coconuts but taking pharmaceutical drugs for all time
02:49.47carrarthats not so bad
02:49.52carrarrelaxed life
02:49.52dandate2haha i know
02:49.56dandate2thats why i like it here
02:50.07dandate2even if i was broke i could live for free off coconuts and bananas and find free housing
02:50.31dandate2all the barong barong here have big screen TVs, and internet service lol
02:50.38carraryeah, if bear can live off piss, you can live off of coconuts
02:50.43carrarheh
02:50.59dandate2but man this place is poor
02:51.05carrarparden my fowl language
02:51.10dandate2i can buy 12 packs of cigarettes for $4 usd
02:51.19carrarbbl, dinner
03:05.35EnrGEso ppl goin to astricon next month?
03:06.01russellbi am!
03:06.23EnrGEthat makes two of us. :) anyone else?
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03:16.48[TK]D-Fenderphix: IP-30X is functionally inferior to SPA-94X
03:17.16[TK]D-Fenderphix: IP32X/33X bring the bar up there that for most use I'd rather have the Polycom.
03:19.10drmessanoAstricon is for those people still using Asterisk.. I have moved on to Facebook-Over-IP
03:19.49KavanSI myspace'd over IP once
03:20.21[TK]D-FenderKavanS: Not in here, this is a FAMILY CHANNEL!
03:20.28KavanSlol
03:20.38p3nguinPfft.  IP is so 20th century.
03:20.50KavanSso when is ipv6 coming to the interwebz?
03:21.03KavanSor is ipv6 going to be vaporware like duke forever
03:22.38KavanSdandate2, that's awesome....12 packs of cigs for $4 usd
03:22.44KavanSdandate2, wtf do you do there to stay alive?
03:22.53[8none1]http://penrose.uk6x.com/
03:23.25KavanSahhh
03:24.41denonKavanS: IPv6 is here .. you're just not in the loop :)
03:24.48denonor should I say Linked In, with the current conversation
03:24.49denon;)
03:25.13KavanSyeah I know it's around...but it's not like your residential isp is issuing ipv6 to customers
03:25.19KavanSunless I was unaware of this?
03:25.28p3nguinThere are some.
03:26.16denonthere are a few yeah
03:27.20denoncomcast claims they're going to do general deployment next year
03:27.23denonand is already doing trials
03:29.08KavanSwow, that's impressive
03:31.31denonspose so
03:31.44denonendlusers wont know any different, except that their firewall won't be as well tested under 6
03:32.00denonso we'll see a new wave of exploits while vendors trying to figure out what a : is
03:35.07carrarrussellb
03:35.14carrarwatch out for swineflu!
03:35.21carrarI'm suppose to go too
03:35.33carrarnot sure I want too after hearing about that other geek convention
03:35.58russellbit would be totally worth it
03:36.13carrarI went 2 years ago, same place
03:36.18carrarwas worth it then
03:36.33russellbah, 2 years ago was Phoenix, but a different venue
03:36.37carrarswine flu is worth it is what you're saying? :)
03:36.42russellbyes!
03:36.44carrarhaha
03:37.04carrarYou'll come to my funeral?
03:37.28carrar"He was a great attendee"
03:37.42carrarkept quite
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04:14.07*** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net)
04:16.28geneticxhello everyone..I would like to get some suggestions. I was given the task of implementing a voip solution in suriname, what would be the best: buying a digium card like a TDM400P and using regular local pots, or trying to find a VoIP provider that has a close POP to reduce latency...hummm what you guys think?
04:17.28MangoIs this for local calls or international?
04:19.21geneticxwell, they are going to be using the digium card for local calling only..as far as international calling which they need, I'm thinking to peer with the asterisk box that we have here in the U.S. and route their international calls..what do you think?
04:20.07MangoYou could peer with the box in the US for North American calling.  Or, direct to the carrier may be an even better option.
04:20.35MangoFor local calling, the less expensive option would likely be to find a VoIP provider with a close POP.
04:20.38geneticxdirect to their local carrier?
04:21.20MangoHowever, I don't know of any quality VoIP providers in that part of the world.
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04:21.43MangoBut if there are one/a few that is the route I would prefer.
04:22.05geneticxyeah that would be ideal, but damn I'm with you on not knowing the quality of VoIP providers there..
04:24.11MangoIt would take some experimenting I imagine.
04:25.01geneticxsince I don't really know any VoIP providers around there, I would probably prefer for them to get a regular analog line to the office for local calling, and implementing international calling somehow either with peering with the asterisk in the states, or just like you said straight to the carrier
04:25.43geneticxyeah, but don't know if I would have time to experiment =D
04:25.55MangoLol.  In that case, ya, a FXO card would likely be your best bet.
04:26.02MangoI've heard good things about Sangoma.
04:26.15MangoYou may want to investigate them as well.
04:26.39geneticxOk, sounds good enough
04:26.59geneticxI'm out, got to do lots of planning tomorrow.. =D
04:27.14geneticxthank you for your advice.
04:28.28Mangogood luck!
04:29.16geneticxthanks! you will probably see me here again as the project advances..=D
04:37.15Gokee2Hello everyone, Is there any way to see the results of a register command?  Should I be showing some stuff in "sip show registry"?  I found a few posts on the mailing list about the sipgate register command not working and that causing incoming calls to not work.  Also I noticed if I command out the register command nothing changes, outgoing still works incoming doe not.
04:39.00[TK]D-FenderGokee2: If you don't give them an IP and you don't register then you aren't going to get calls.
04:39.06[TK]D-Fender~sipregister
04:39.06infobot[~sipregister] SIP Registration is to tell your provider as to what IP address & EXTEN to send INCOMING calls to. Some ITSP's let you use a fixed IP or host rather that register.  Registration is NOT normally needed to PLACE a call, as those are typically auth'd independently.  Others accept UN-AUTH'D calls once you are registered (saves on negotiation BW)
04:40.20Gokee2[TK]D-Fender, I have a register command I just don't think its working...
04:40.57[TK]D-FenderGokee2: And what does "sip show registry tell you"?
04:41.17[TK]D-FenderGokee2: And what does "sip show registry" tell you?
04:41.34*** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id)
04:41.53*** join/#asterisk geneticx (n=geneticx@adsl-2-59-65.mia.bellsouth.net)
04:42.22Gokee2[TK]D-Fender, Wow, after a few days of fighting with it I just realized something...  My register command was not in the general section!  I changed it and now sip show registry shows stuff
04:43.33*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
04:43.35Gokee2And now incoming calls work!
04:43.53[TK]D-Fender\o/
04:45.07superbeefIf i wanted a sandbox enviornment, would it be possible to have 2 PBXs and have a T1 card in each connected direclty to each other?
04:45.34Gokee2My, I feel really stupid wasting days over that little problem
04:45.37denonsuperbeef: of course
04:46.08superbeefcool
04:46.22superbeefi need to do that.. i'm getting burned switching from zaptel to dahdi with a sangoma card
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05:04.47m477autrying to initiate a call from the asterisk manager
05:22.28tzafrir_laptopAction: Originate
05:29.39m477auyeah
05:29.44m477ausorry, forgot I started typing something
05:29.49m477aukeep getting this in the logs
05:29.52m477au== Starting SIP/6000-0da30a60 at from-internal,SIP/iinetout/0402686828,1 failed so falling back to exten 's'
05:30.00m477authat's for internal to external
05:30.14m477auhowever if I call something on the iinetout trunk to an internal extension
05:30.15m477auit works
05:30.56m477aufputs($socket, "Action: Originate\r\n" );
05:30.57m477au<PROTECTED>
05:30.57m477au<PROTECTED>
05:30.57m477au<PROTECTED>
05:30.57m477au<PROTECTED>
05:30.57m477au<PROTECTED>
05:34.14kaldemarm477au: SIP/iinetout/0402686828 is not an extension
05:35.10m477aueven with just 04 it does the same thing
05:35.50m477auhowever, if I change it to *97
05:35.54m477auit connects me to voicemail fine
05:37.44kaldemarm477au: the exten needs to be a matching extension in from-internal
05:42.34m477auok, so I need to set up a context for this to work as desired
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05:44.41Gokee2Whats the difference between TDM402B and the  TDM410P?  I also see "Wildcard" from time to time, whats that mean?  Anyone know of a list of all the digium cards?
05:45.17kaldemarm477au: no, you need to call the right extension
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05:48.42kaldemarGokee2: give links that refer to those, since digium doesn't have products by those names
05:49.49m477aukaldemar, I'm not calling an extension, I'm calling an external number
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05:50.36kaldemarm477au: with originate, you call an extension. extension is a number in your dialplan.
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05:54.59Gokee2kaldemar, For the TDM402B http://www.cetusvoip.com/product_info.php?products_id=1801 and http://www.soho-voip-phone.com/Asterisk_Hardware/digium_Digium_TDM402B.html wildcard I have seen http://www.ipphone-warehouse.com/Digium-Wildcard-TDM02B2-p/tdm02b.htm  I forget where else I have seen wildcard but I know I have seen it somewhere else as well
05:56.32kaldemarGokee2: that TDM402B has 2 FXO modules.
05:57.06m477aukaldemar: found the problem, lack of callerid
05:57.18m477au<PROTECTED>
05:57.19m477au<PROTECTED>
05:57.19m477au<PROTECTED>
05:57.49m477ausoon as I added a callerid, it worked.
05:57.58kaldemarGokee2: that TDM02B2, based on the picture, is a discontinued analog card model.
05:59.39Gokee2kaldemar, Ah, I guess I should find a non-discontinued card to buy then
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06:00.08kaldemarGokee2: wildcard is just a name part in some models.
06:00.27kaldemarGokee2: look at the ones listed here: http://www.digium.com/en/products/
06:02.16Gokee2Why do I see some TDM410p's?  All thats listed there (for analog 4 ports) seems to be the TDM410?
06:03.06kaldemarwhat do you mean?
06:03.46Gokee2Like say here http://cgi.ebay.com/Digium-TDM410P-with-2-FXO-ports_W0QQitemZ300346351007QQcmdZViewItemQQptZLH_DefaultDomain_0?hash=item45ee099d9f&_trksid=p3286.c0.m14 lits it as a TDM410P
06:04.43kaldemarGokee2: so what? it is a TDM410.
06:05.00Gokee2kaldemar, Then whats the "P" for?
06:05.05*** join/#asterisk tm1985 (n=tm@082-146-101-077.stat.adsl.xs4all.be)
06:05.27kaldemarTDM410 is the card itself. you can equip it with modules: http://www.digium.com/en/products/analog/s400m.php
06:05.44kaldemarresellers call their combinations whatever they wish
06:05.50Gokee2telephonydepot.com also lists it as a TDM410P?  Can I get a non-P version?
06:06.10Gokee2(telephonydepot.com has no modules)
06:06.22kaldemarGokee2: there's no P in TDM410. completely different digital cards have P in their model.
06:06.55kaldemarit's just a letter they chose to put in an item. it means nothing.
06:07.03Gokee2kaldemar, Ok, I won't try and make since of all the different stuff at the end of the card name then
06:07.59kaldemardon't look at the model names, try to figure out what they're actually selling you instead.
06:08.24Gokee2ok
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06:43.44tm1985can anybody help with a dahdi issue?
06:50.37tm1985when I got a incoming call it gives for example 13121110 on my phone display but I need to have a 0 in front of them so that I get 013121110
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06:52.39kaldemartm1985: modify callerid in your dialplan. Set(CALLERID(num)=0${CALLERID(num)})
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07:03.05superbeefIs this invalid in asterisk 1.4?   exten => _.,2,Goto(from-internal|BYEXTENSION|1)
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07:04.36tm1985kaldemar, when I do that I want to call a queue then I get moh and no ringtone
07:04.39*** join/#asterisk gego (n=rick@b238085.customer.hansenet.de)
07:04.50kaldemarsuperbeef: BYEXTENSION hasn't been valid for years
07:05.08superbeefkaldemar: lol... context rules from really old asterisk installs
07:05.17superbeefi just rebuilt one of them,, now it's not playing nice
07:05.19kaldemarsuperbeef: replace it with ${EXTEN}. and replace _. with something else, e.g. _X.
07:06.01superbeefit has _X my paste was just bogus
07:06.36superbeefso.....    _X.,2,Goto(from-internal|${EXTEN}|1)
07:06.57kaldemartm1985: so you wan't ring tone? "core show application Queue" in CLI will give you options for app Queue. you'll find the answer there.
07:07.28tm1985no I need a ring tone but I don't get it?
07:08.00kaldemars/wan't/want/
07:08.16kaldemartm1985: read that again
07:08.24tm1985oke
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07:10.19superbeefkaldemar: hey thanks, that fixed it.. now I can sleep!
07:10.38tm1985kaldemar thx found it
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07:14.32tm1985kaldemar if you dial with DAHDI you need for example Dial(DAHDI/1/${EXTEN}). Can you change the 1 from channel with a asterisk variable like the ${EXTEN}
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07:15.58kaldemartm1985: sure
07:16.06tm1985what is it called
07:17.00kaldemaryou're making no sense now. you need to define a channel in the dial. there is no channel until you do so.
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07:17.32kaldemarif you want the current channel, it is stored in CHANNEL, but you can't dial the same channel you're already using.
07:19.01kaldemarwhat you might be looking for is groups in chan_dahdi.conf. you can assign multiple channels to a group and then use for example Dial(DAHDI/g0/${EXTEN}) to dial using some available channel in the group.
07:20.06tm1985ah I will try this
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07:32.09datacompboyHi all! :) Is there method to prefix dialed extension from particular peer ?
07:32.49datacompboyi.e. all calls from [peer1] come to [general] section as-is; but all calls from [peer2] come to [general] as p-${EXTEN}
07:33.12datacompboymain problem is extensions not numbers, but alpha-numeric
07:33.26tm1985kaldemar thx that is what I needed
07:35.14kaldemardatacompboy: make an own context for peer2 where you add the prefix and the go to [general]
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07:35.31datacompboykaldemar: well, how to match any alpha-numeric extensions,
07:35.48datacompboy? if i do exten => _., -- it match also s, h internal extensions
07:37.12kaldemardatacompboy: _[a-g,jr] <-- like that
07:37.28kaldemarso don't do _.
07:38.07datacompboybut what if incoming call will be to s@gate ? how to separate internal "s" and incoming "s" ? that is main problem
07:40.30kaldemarwith different contexts
07:41.03kaldemaror just give yourself a break and don't allow arbitrary alpha-numeric extensions
07:41.37datacompboykaldemar: that are SIP<->Skype gate. so ther will be arbitraty alpha-numeric (and dots, and etc) extensions...
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07:45.05MiccIs there a way to park a call in a specific parking spot instead of using the park exten which reads it back to you?
07:45.36tm1985kaldemar do you know something of redirect calls to mobile numbers with dahdi?
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07:47.16kaldemartm1985: please elaborate
07:49.37tm1985We have a isdn provider an we are possible to redirect calls to for example a mobile phone by dialing *21*TELNR#
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07:50.44tm1985Now I want to do exten => 600,1,Dial(DAHDI/g0/*21*telnum#) but that doesn't work do you know how I have to do that
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07:52.57user4545Hi, i have a problem... if I call to my number by Sipgate, then I become message in asterisk: "Call from '1234567' to extension '1234567' rejected because extension not found."
07:53.02user4545why?
07:53.10user4545can any me help?
07:53.27tm1985Do you know something about that?
07:54.12kaldemartm1985: well, not unless you tell how it doesn't work.
07:55.56kaldemaruser4545: you need to add the extension in your dialplan.
07:56.19user4545I have it extension
07:56.29tm1985http://pastebin.com/d50aed328 here are my errors
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07:57.04tm1985http://pastebin.com/d5a159049
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07:57.12tm1985that are my exten for the 600
07:57.47user4545kaldemar: I have it extension in my dialplan
07:58.06kaldemaruser4545: not in the right context, show your dialplan and the relevant sip.conf context
07:58.15tm1985http://pastebin.com/m5ebc2c38 => exten
07:59.07kaldemaryour cli output doesn't match the dialplan
08:01.43tm1985The wierd thing is when i call a sip are playback a soundfile It that exten works but when want to dial that *21*TELNR# I get the error that extension isn't know
08:02.01user4545kaldemar: http://pastebin.ca/1568049 my dialplan please see it
08:10.41tm1985Hello need some help with DAHDI. I have provider that provides ISDN and not VOIP. Our provider give us also to chance to redirect to for example a mobile phone with *21*TELNR#
08:11.58tm1985I'm trieing to dial this with exten 600 but always get this error : http://pastebin.com/d50aed328
08:12.05tm1985can someone help me???
08:13.09tm1985http://pastebin.com/d50aed328 this Is that exten 600
08:13.56tm1985sorry this is the exten 600: http://pastebin.com/m50b7a4d4
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08:18.17tm1985is nobody here who can help me?
08:25.02tm1985Hello need some help with DAHDI. I have provider that provides ISDN and not VOIP. Our provider give us also to chance to redirect to for example a mobile phone with *21*TELNR#
08:25.13tm1985I'm trieing to dial this with exten 600 but always get this error : http://pastebin.com/d50aed328
08:25.25tm1985can someone help me with this???
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08:34.27Polysicshello
08:34.38Polysicsi finally managed to fix the sip users setup
08:35.05Polysicssince i am using mysql for the sip users, do you recommend using db for the extensions too?
08:35.08tzafrir_laptoptm1985, what is '*21*' ? A valid number?
08:35.22Polysicsit is going to become a 200ish users distributed service
08:36.01tm1985What do you understand under valid Number?
08:36.38Polysicsso far everything is working with static extensions
08:37.33Polysicsbtw, if i put single user extensions in db, can i still use static extensions for some things?
08:40.16tm1985http://pastebin.com/m6b378094
08:40.36tzafrir_laptoptm1985, you try calling that number . Is it a number you can actually dial to?
08:41.22tzafrir_laptop*21*TELNR# is not a valid phone number
08:42.08tm1985no there we have set our number that we want to dial but I change that so that not everybody has my personel mobile number
08:42.15tzafrir_laptopYou probably meant *21*83567
08:42.21tzafrir_laptoperr...
08:42.27tzafrir_laptopYou probably meant *21*83567#
08:42.39tm1985yes
08:42.40user4545Hi, i have a problem... if I call to my number by Sipgate, then I become message in asterisk: "Call from '1234567' to extension '1234567' rejected because extension not found."
08:42.59tzafrir_laptopI also suspect that '#' is not part of the number
08:43.00user4545but I have this extenstion
08:43.36tzafrir_laptopDoes Sipgate have this extension?
08:43.37tm1985yes for redirecting to that number you have to dial *21*83567# for example
08:43.40wathekany one would help me to test my Asterisk configuration please ?
08:45.07tm1985the # is needed tzafrir!!!
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08:45.41user4545<PROTECTED>
08:46.03user4545see it
08:46.05user4545http://pastebin.ca/1568049
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08:50.27user4545can anybody help me?
08:50.27tm1985tzafrir the # is needed in the phone number
08:52.43mort_gibCrackling noises on incoming call but not outgoing?? Any ideas??
08:53.32Polysicsdoesn anyone used realtime extensions with mysql?
08:53.53Polysicsif i move them there, will the extensions in leave in extensions.conf still work?
08:56.30tm1985Does anyone knows how I can dial *21*TELNR# with dahdi?
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09:08.38renzoeguys, can you enlighten me on what PRI line is? the digium reseller is recommending us to have PRI for 60-70 office users
09:08.49tzafrir_laptoptm1985, please enable pri debug an show a trace of the attempted call
09:09.22wonderworldrenzoe: PRI is a multiplexed ISDN. you get about 30 ISDN channels from one line, depending on where you live
09:09.40wonderworldi think in the US it's 28 channels
09:09.46wonderworldin europe 32 or so
09:10.05renzoeis this PRI can have a unique telephone number per ip phone?
09:10.11renzoehere in the UAE they said 30
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09:10.14tm1985how do you enable that debug tzafrir?
09:10.16wonderworldso you wouldn't need to book 30 ISDN-lines for your 60 office users
09:10.52renzoe30 channels meaning 30 simultaneous inbound/outbound?
09:10.53wonderworldyou normaly get a 100-number number-block like 1111111-0 to 1111111-99
09:11.07wonderworldrenzoe: yes, 30 "real" ISDN lines
09:11.33wonderworldyou have to check the details with your telco. no idea about how things are in the UAE
09:12.24renzoebecasue we are thingking of uilding our own asterisk based system the first one we checked is setting up 8 ports digium AEX800 cards with echo canceller. which do you think is the mest setup?
09:12.47renzoe*best
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09:13.26wonderworldnope, the AEX800 card is a card for analogue phones
09:13.37wonderworldyou need a card that supports E1/T1
09:13.39renzoewonderworld. yes he told me that also that i will get 100 numbers
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09:14.32wonderworldhttp://www.voipsupply.com/dgm-te205p
09:14.42wonderworldthats a digium card for T1 lines
09:14.48renzoeso a cisco ip phone will not be compatible with AEX800? really sorry just new to this pbx thing
09:15.07Chainsawrenzoe: Cisco IP phones work on ethernet.
09:15.26wonderworldyes, you first need to connect the asterisk-pc to your telco somehow
09:15.29Chainsawrenzoe: The only telephony adapter you need in your server is one to connect to your ISDN line(s).
09:15.41wonderworldhow you connect your phones in your company with asterisk is a different thing
09:16.21renzoei see. now getting some light
09:16.39wonderworldlike [telco providing T1 line ] --> [Asterisk ] -> [LAN] -> [Your phones]
09:16.54renzoe:) so i need to accomplish forst on how to connect my asterisk to my telco
09:17.10wonderworldprobably yes :)
09:17.52wonderworldcheck if your cisco phones can do SIP.... that would be one standard wy to connect them to asterisk
09:18.18renzoebut will the AEX800 can do the thing? we also in a tight budget. if either can do the same thing then the next think i will look is the cost
09:18.32tm1985http://pastebin.com/m8fd7417
09:19.26wonderworldnope. the AEX800 is just for connecting analog telephone lines or analog phones to asterisk. your T1 lines isn't analog and your Cisco phones aren't either.
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09:19.59tm1985Does anyone knows how I can dial *21*TELNR# with dahdi?
09:21.14wonderworldtm1985: never did that, what happens when you just dial it with dahdi?
09:21.17renzoei see got it. by the way how much is the PRI in your area? i already sent an inquiry but it really takes time to reply since its ramadan here
09:21.37wonderworldi live in germany and i can get a PRI for €99 a month
09:21.48wonderworld(just for the line, no calls included)
09:22.18tm1985It doesn't work don't know how I have do that in asterisk and dahdi
09:22.35renzoeis that euro?
09:22.38tm1985And can't find any information also
09:23.03wonderworldyes, EURO
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09:23.57wonderworldtm1985: what do you want to do with that code?
09:24.22renzoethanks a low wonderworld. now i just waiting for the supplier to reply
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09:24.40wonderworldno problem....
09:24.51tm1985to redirect calls to that number
09:25.28wonderworldwhere is the call coming from?
09:25.43tm1985outside
09:26.05wonderworldand it sould go to another outside number?
09:26.27tm1985indeed
09:26.32wonderworldok
09:26.49wonderworldi don't really get why you would need the *21*123#
09:26.59wonderworldwhy not let dahdi just dial out to that number?
09:27.20wonderworld* would switch the calls together
09:27.22renzoeby the way wonderworld, once the PRI card is installed its already capable of call listen/intrude and recording? or asterisk is the one handling it?
09:27.46wonderworldasterisk is doing that
09:27.52wonderworldlook into chan_spy and mixrecording
09:28.17wonderworldbe sure to buy a card with a hardware echo canceler too
09:29.30renzoeis that an add-on to the PRI card? or a separate PCI card?
09:29.35tm1985to redirect automaticlly in the weekends
09:30.18wonderworldtm1985: you can do that with dialplan logic if your approach fails
09:30.25renzoei saw the echo cancellation in digium and its an add-on to the aex800
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09:31.35mbrevdais seeking uk businsess grade did's
09:32.45wonderworldtm1985: http://www.voip-info.org/wiki/view/Asterisk+tips+openhours
09:34.06wonderworldrenzoe: thats an add-on to the pci-card. some merchants sell the card with it some sell it without
09:34.17wonderworldbe sure to get one with EC, because you will need it
09:34.35wonderworldasterisk can do software EC but it doesn't work as well as a hardware solution
09:34.44wonderworldyou will have echos if you don't get a hardware EC
09:37.16wonderworldto save you a lot of time and hassle ask your telco as well about their signalling settings (you'll have to put them into the config-file for your pci-card)
09:46.50tm1985Does anyone knows how I can dial *21*TELNR# with dahdi?
09:47.14tm1985this for redirection to a mobile phone
09:48.08*** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com)
09:50.12tzafrir_laptoptm1985, for starters, 'T', 'E' etc. are not digits you can dial
09:50.57tm1985for TELNR we use a number like 049856862 for example
09:51.46tm1985we don't use *21*TELNR# we use for example *21*045632869# for example
09:52.33wonderworldtm1985: as i told you, for the thing you want to do http://www.voip-info.org/wiki/view/Asterisk+tips+openhours would probably just work fine
09:53.51tm1985But we want to set it one when the last one leaves and back off when the first person come back we can't predict when does hours are exactly
09:54.40wonderworldok, then you can create an extension like 9999 or something that just writes a variable to the asterisk DB like "0" for normal office hours and "1" for night service
09:55.05wonderworldyou can read that var in your extension and send calls to the right place afterwards
09:55.27wonderworldlike people can call "9999" on their phone when they leave
09:55.37wonderworldand calls will be rerouted after that
09:58.06tm1985That is not what we really looking for
10:00.18wonderworldk then i didn't get you, sorry
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10:17.37cjkhi, what is the best way to disable musiconhold for zap channels?
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10:22.32[netman]hi all, I got an "Unable to support trunking .... without zaptel timing" error , but my ztdummy module is loaded. Any suggestions, please?
10:25.12angryuser[netman], type "zaptel show status" in CLI look if it really On
10:25.24angryuserit is*
10:26.38[netman]thx angryuser , give a minute, plz
10:26.39garymcHi I altered some default passwords, couldnt get into my gui. Sorted that now. But my 2 phones show as connected, but they cant call each other anymore. Here is an asterisk output. http://pastebin.ca/1568116 Can anyone tell me whats going on here?
10:27.14wonderworldcjk: do you use the m option in your dial command?
10:27.42kaldemargarymc: go to #freepbx
10:27.59[netman]angryuser: that is. Thank u very much
10:28.01cjkwonderworld, no
10:28.06garymcThought it coulda been an asterisk thing Kaldemar
10:28.17garymcive put it in there also :S
10:30.46garymcIts ok i fixed it
10:30.49garymcsorry
10:31.26wonderworldcjk: for whom do you want to disable it? people calling you from outside to a sip-phone inside?
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10:56.35garymcanyone got a minute to test my sip extension?
10:56.45garymcusing zoiper or some other softphone?
10:57.02garymcjust opened ports you see :S
11:01.57tm1985Does anyone knows how I can dial *21*TELNR# with dahdi?
11:03.04*** join/#asterisk Intel`` (n=clc@213.132.40.2)
11:03.22Intel``hi guys, which do you prefer. setting up fxo or setting up isdn?
11:06.51cjkwonderworld, people from the outside calling activate my moh which i hear instead of theirs
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11:09.05garymcim gonna be setting up isdn so ill choose isdn :P
11:09.08tm1985Does anyone knows how I can dial *21*TELNR# with asterisk and dahdi?
11:16.37Intel``garymc what are the avantages? i dont know if isdn will be too much for our requirements
11:17.23garymcISDN30 is cheaper for me than a good broadband connection. I can get 30 channels down one cable
11:17.46garymcBut at the minute im only gonna have the smallest option and thats 8 channels
11:18.21garymcso its like £300 per quarter. But if i want a 2mb up and down broadband they want £1200 per quarter
11:18.39Intel``by the way im new to pbx and i wanted to know the meaning of "channel" and "lines" because usually that's the first one sales are asking
11:19.10Intel``they were asking me how many
11:19.33garymcchannels are calls down one line
11:20.30garymceg. I got ISDN30 with 8 channels. ( ican have 30 channels if want but need to pay more per quarter) this means i can have 8 simultaneous calls at the same time down the one line
11:20.50garymcso i can have 8 workers on the phone to differnt customers at once
11:21.19fiddurtm1985: You keep asking the same question.  If people doesn't answer, try putting the question in a different way...  Maybe you just want   exten => _*21*X!#, 1, Dial(DAHDI/g0/${EXTEN}) ?   dialling * and # really shouldn't be any different from any number.
11:21.50Intel``channels are i see i see so if i will get 10 lines with lets say 8 channels each, that's 10*8 users in a call simultaneously?
11:23.04garymcyeah thats 80 ncalls at the same time
11:23.06garymcI think
11:23.12garymc*calls
11:23.16Intel``yup
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11:24.09Intel``but can i get 1 line with lets say 20 channels? will that get clooged up or something?
11:25.21Intel``that's why you distribute it on multiple lines?
11:25.44tm1985Has anybody ever just to dial *21* in asterisk?
11:26.29fiddurtm1985: Yes, that's not a problem.
11:27.40tm1985with that exten that you entered above?
11:29.05tm1985Because that don't work with me!!!
11:29.15fiddurtm1985: JUST *21* is just Dial(DAHDI/g0/*21*)   ...but the provider then checks if that is a valid number or not.  Most probably, it will answer that it's not a valid numer, e.g. hangupcause 28
11:29.58fiddurtm1985: If the provider wants *21*xxxxxxxxxx#, then you have to include the whole number, not just *21*
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11:30.04doolittleworkhi htere
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11:31.06fiddurtm1985: lines 6-7 in your paste http://pastebin.com/d50aed328 shows that the provider doesn't like the number *21* in itself.
11:31.07tm1985I have set exten => 600,1,DIAL(DAHDI/g0/*21*0489562356#) but that doesn't work
11:31.54fiddurtm1985: In the paste that's not the exten used...  maybe you missed a 'dialplan reload'?
11:32.30doolittleworki want to record my sip channel if i use the mixmonitor works fine, but once i end the call it ends the monitor and end the call, how can i get monitor called played back 2 me once i end the call?
11:32.46tm1985If i do anything else but the *21* dial it see the exten 600 and use it
11:34.10wonderworldtm1985: maybe you have to pause for a short amount of time after the *21* ?
11:34.36wonderworldfor your telco to recognize it?
11:34.42fiddurtm1985: In http://pastebin.com/m8fd7417  it states clearly that "Ext: 1  Cause: Invalid number format (28),"
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11:35.18fiddurtm1985: I think this is really a question for your provider, not a question of how to get asterisk to send the number.
11:35.39tm1985aahhh oke
11:36.46tm1985I will try with a pause
11:37.06wonderworldi wouldn't know how to do it though
11:37.47Intel``what's the difference between free and commercial asterisk softwares?
11:38.05wonderworldi think on cell-phones you can hold the "0" key for a while and it will put a "p" into the number. that does pause a second or so. not sure how to do that with asterisk.
11:38.32doolittleworki want to record my sip channel if i use the mixmonitor works fine, but once i end the call it ends the monitor and end the call, how can i get monitor called played back 2 me once i end the call?
11:38.51fiddurtm1985: From Dial: "If you need a .5 second pause while dialing a number you can insert a w in the appropriate place. "
11:39.55fiddur...but since they said invalid number for *21* I doubt that will work
11:39.59Intel``guys. is call monitor,barge,whisper not available on asterisk based pbx? just digium appliance?
11:40.12tm1985I doesn't work
11:40.31Intel``i am reading a comparison and the supplier tells me that its not. but just want to confirm
11:40.55kaldemarIntel``: yes they are
11:41.08wonderworldi am having a problem with transfering calls. i assigned *1 and *2 in features.conf to enable people to transfer calls. it works, but not in a reliable way. every 5th try or so fails. they hear "transfer" they enter the extension but the call is never transfered. any idea how to debug that?
11:41.39wonderworldi played around with the timeouts and set them real high for testing but it didn't fix it
11:41.58tm1985fiddur My provider offers the *21*NUMBER# for redirecting calls
11:42.11wonderworldphones are SNOM hardware sip-phones
11:42.14*** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net)
11:42.41Naikrovekyawns
11:43.10fiddurtm1985: Well, from your logs they reject that particular number in your particular case.  You'd better ask them.
11:43.40tm1985when I use a phone outside asterisk and dial *21*NUMBER# it works
11:44.56wonderworldtm1985: try to use it with the redial-function of that "outside" phone and check if it still works.... to verify that it's not about pausing after the special combination
11:46.45kaldemartm1985: what phone are you using outside asterisk?
11:46.57kaldemartm1985: are you using the same ISDN line?
11:47.13tm1985yes
11:48.03tm1985we are using the samen isdn line
11:48.33kaldemarwith an ISDN phone?
11:49.13tm1985In asterisk i'm using IP phone SPA962
11:49.18fiddurtm1985: And normal calls from asterisk to other numbers work on DAHDI/g0?
11:49.29tm1985yes
11:49.55wonderworldmaybe it's about "pridialplan" in chan_dahdi.conf
11:50.27fidduryes, it could be dahdi that says invalid number format, without even sending it to the provider perhaps...
11:50.41fiddurI'm not enough versed in dahdi i'm afraid...
11:50.54wonderworldme neither, just guessing
11:50.58wonderworldcheck if thats set to unknown
11:51.02tm1985what do you have to do with pridialplan
11:51.24tm1985yes but it's set into comment
11:52.08tm1985does it need to be unknown or not?
11:53.25wonderworldi way really just guessing. if it's commented out, it probably already is unknown as that seems to be the default
11:54.09wonderworldyou might want to try to play around with other values for pridialplan and prolocaldialplan
11:54.10fiddurtm1985: pridialplan default is national according to http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf
11:54.25wonderworldnot sure if it would do anything good but worth the try
11:54.56wonderworldyou have to do a "dahdi restart" in cli after doing changes to chan_dahdi.conf
11:55.09wonderworldwatch the cli, i think for some changes to dahdi you even have to restart asterisk
11:59.06tm1985I tried national and unknown but result stays the same
11:59.46wonderworldk, now i am really out of ideas..
11:59.56wonderworldmaybe try posting your problem on some forums.
12:00.05wonderworldor ask again later
12:00.22tm1985I put it on several forums
12:01.05tm1985I searching for several days now
12:01.08wonderworldwhat happens when you dial the number from a phone atatched to asterisk instead of dialing it automatically from the dialplan?
12:02.18wonderworldi know, should be exactly the same thing, but who knows....
12:02.40tm1985I type *21 and then I get that there is no extension *21
12:03.54wonderworldcreate one for it and try again
12:04.02tm1985Call from '40' to extension '*21' rejected because extension not found.
12:04.20*** join/#asterisk bboness (n=bones@acdc.internet.ao)
12:05.02wonderworldlile exten => _*21X.,1,
12:05.22bbonessIs there any way to define the source ip address when talking to a sip peer?
12:05.30*** join/#asterisk mwalling (i=mwalling@97.107.128.165)
12:05.38fiddurtm1985: If you call *21* from a normal phone, is it answered first before you give the rest of the number + # ?
12:06.00tm1985fiddur yes
12:07.20fiddurtm1985: You could try Dial(DAHDI/g0/*21*,,D(0489562356#))  perhaps... then it waits until answer, rather than waiting exactly .5 seconds...
12:08.13wonderworldfiddur: THAT sounds good
12:08.18fiddurbut still.. since *21* gave hangupcause 28, I don't see that it could work...
12:10.02tm1985I get the same problem!!!!
12:11.48tm1985exten => 600,1,Dial(DAHDI/g0/*21*,,D(0498506822#))
12:11.58tm1985In extensions.conf
12:12.06wonderworldcause 28?
12:12.09*** join/#asterisk AllstateComputer (n=brian@c-76-108-186-218.hsd1.fl.comcast.net)
12:12.41*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
12:12.59*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
12:13.50tm1985http://pastebin.com/m225855ca yes again cause 28
12:15.42wonderworlddid you try a PRI debug already?
12:16.28*** part/#asterisk mwalling (i=mwalling@97.107.128.165)
12:16.29wonderworldto see what is going on exactly?
12:17.04wonderworldis it a digium card on ptp-ISDN ?
12:17.06tm1985yes set that in the pastebin
12:17.26tm1985<PROTECTED>
12:17.35tm1985and use zaphfc
12:17.44wonderworldk. you might want to try misdn instead of dahdi
12:18.27wonderworlddahdi sucks with european telcos and isdn. many features arent't properly included for the european market
12:18.39wonderworldor play around with chan_dahdi.conf
12:18.48wonderworldi think thats really all i can suggest
12:19.50tm1985I have tried isdn with mISDNv2 same problem
12:19.55wonderworlddoh
12:20.23tm1985Somebody suggest to try it with dahdi
12:21.27wonderworldor try to get to some technician at your telco. the standard hotline won't probably know what to do
12:21.55fiddurSince normal calling works, it is obviously working...  just weird for this special number...
12:22.17tm1985I know it sounds wierd
12:23.29*** join/#asterisk ariel_ (n=chatzill@63.214.236.169)
12:23.42wonderworldor could it be that your "normal" working phones send something else than *21* when you dial *21* ?
12:23.44*** part/#asterisk Grof (n=dule@89.201.165.226)
12:24.04*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
12:24.04*** mode/#asterisk [+o leifmadsen] by ChanServ
12:24.04wonderworldwell probably not if thats the code you got from the telco...
12:24.27tm1985I think not
12:24.32fiddurtm1985: do you have support on the isdn-card?  ...you could try asking them...
12:25.21tm1985I goin to restart my the program
12:25.41ariel_Morning everyone
12:26.09leifmadsenmorn
12:26.42ariel_does anyone know of a way via asterisk 1.6 to resend to the phones the message that they have voicemail?
12:26.45*** join/#asterisk tm1985 (n=tm@082-146-101-077.stat.adsl.xs4all.be)
12:26.54wonderworldariel: yes
12:26.56leifmadsenariel_: MinivmMWI
12:27.12tm1985wonderworld I had problem with my chat program
12:27.21leifmadsenthere are possibly other ways of doing it
12:27.24ariel_minivmMWI wow, don't know this one will look it up t/y
12:27.46ariel_I have some polycom's 8020/8030 that are passive phones
12:28.17ariel_they get the vm icon if there on but if there turn off and back on they don't get it. until a new vm comes in.
12:28.18*** join/#asterisk magronez (n=eusei@unaffiliated/magrao/x-2903)
12:28.22leifmadsenariel_: I used it to control MWI for hot-desking agents
12:28.36leifmadsenariel_: then they need to SUBSCRIBE to a mailbox
12:28.38*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
12:28.42ariel_leifmadsen: t/y
12:29.00leifmadsen(SUBSCRIBE is caps because that is the SIP message, and not because I'm yelling at you like [TK]D-Fender)
12:29.00tm1985If you have some ideas for the *21* let me know
12:29.23ariel_I know this command
12:29.33ariel_I fully understand been here long enough
12:30.10tm1985Does anyone use *21* to redirect phones????
12:30.12[TK]D-Fenderleifmadsen: focus, not "yelling"
12:30.21tm1985with asterisk and dahdi???
12:30.22leifmadsenI don't agree
12:30.23[TK]D-Fenderleifmadsen: Yelling is all-caps :)
12:31.17*** join/#asterisk gsiener (n=gsiener@d-63-245-116-186.batelnet.bs)
12:31.50[TK]D-Fenderleifmadsen: Half of communication is interpretation and you're certainly entitled to yours.  I have however cleared the reasoning behind the intended interpretation. :)
12:32.18*** join/#asterisk voipmonk (n=voipmonk@dsl-67-212-15-216.acanac.net)
12:32.50leifmadsenwe'll agree to disagree then
12:32.50ariel_wow there is only 4 lines of info on voip-info for mini-vm
12:33.00leifmadsenariel_: there is lots more in doxygen
12:33.11leifmadsenhttp://www.asterisk.org/developers
12:33.24wonderworldwho is running voip-info anyways?
12:34.13wonderworldis it associated with asterisk / digium ?
12:34.35[TK]D-Fenderleifmadsen: I'll see how many disagree along with you after clarification before I openly accept "we" as a substantial percentage.
12:34.37[TK]D-Fender;)
12:35.05leifmadsen[TK]D-Fender: I meant you and me will disagree. I wasn't speaking for anyone else.
12:35.23leifmadsenwonderworld: it is not -- it is run by a third party company. The name is at the bottom of the page.
12:35.38[TK]D-Fenderleifmadsen: Yup.. I clearly haven't had enough coffee yet...
12:35.56[TK]D-Fender~wikis
12:35.57infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
12:35.57NaikrovekI wonder if you'd ever get enough coffee in you
12:36.19Naikrovekhaven't seen you cross the line from cranky to hyper yet
12:36.19[TK]D-FenderNaikrovek: Nope, still a little blood lingering in my caffeine steam :)
12:36.25Naikrovekhehe
12:36.30wonderworldthat's strange. if they go down, 90% of documentation will be gone :)
12:36.50Naikrovekwonderworld: a lot of it is rubbish
12:36.55leifmadsenand 70% of out of date odcs
12:36.56leifmadsendocs*
12:37.01wonderworldyes it is....but it helped me many times
12:37.04[TK]D-Fenderwonderworld: Yes, but 99% of wrong documentation ;)
12:37.11*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
12:37.41fiddurleifmadsen: but easier to find that asterisks real documentation...
12:38.14fiddurleifmadsen: search the net for asterisk application dial, and where do you end up?
12:38.14leifmadsenthe doc/ directory is hard to find, along with the generated PDF in every release?
12:38.28fiddurleifmadsen: Yeps.  Everyone looks at the net first :)
12:38.33leifmadsenfiddur: I just always run 'core show application <foo>' from my console
12:38.56wonderworldhttp://www.the-asterisk-book.com/ is ok as well
12:39.09Naikrovekwho is writing that asterisk cookbook
12:39.16fiddurleifmadsen: Yes... And I know it's there too, but newbies don't... and somehow, I prefer documentation in a browser rather than in a cli...
12:39.20Naikrovekthey have a bunch of recipes on a wiki that can be edited
12:39.33leifmadsenNaikrovek: no one is writing that book. It is dead.
12:39.40Naikrovekd'oh
12:39.43Naikrovekbummer
12:40.08leifmadsenI was one of the ones who was supposed to write it, but then I got busy doing consulting and making actual money :)
12:40.24leifmadsen$500 every quarter doesn't quite cut it
12:40.41Naikrovekfair enough
12:40.45wonderworldyes
12:40.50fiddurleifmadsen: I's appreciate an online version of the built in help, with the usual user-contributed comments etc...
12:41.03leifmadsenfiddur: it'll happen
12:41.05wonderworldi need to make money as well. (plus i wouldn't be able to write an asterisk book)
12:41.05Naikrovekyeah that could be a good idea if it were maintained
12:41.30fiddurautomatically published for released versions off course...
12:41.33fiddur-f...
12:41.36leifmadsenAsterisk applications and functions are already in XML format. You can convert that to HTML with an interpreter/parser
12:42.18fiddurleifmadsen: Yes... and then put up a website, and let users comment it... probably easy to do, but who'll do it? :)
12:42.24leifmadsensomeone will
12:42.36Naikrovekeventually
12:43.05*** join/#asterisk Skeeter- (n=wil_c_wi@c216.218.2-65.clta.globetrotter.net)
12:43.14Skeeter-who wants some money
12:43.15fiddurleifmadsen: But don't get me wrong.  It is a good documentation, and the doxygen code-docs are really good too (at least from the files actually used it :D )
12:43.29*** join/#asterisk icyValk77 (n=icyValk7@gateway.ash.thebunker.net)
12:44.13wonderworldisn't there a web-version of the doxygen docs?
12:44.21leifmadsenhttp://www.asterisk.org/doxygen/trunk/
12:44.23fiddurwonderworld: yes, they are on the web :)
12:44.45fiddurand, they are available for different versions, as the user-docs would be then...
12:44.49Skeeter-pm me if you are interested
12:44.51ariel_mini-vm seems that it might work, but I can't switch the system to that format just yet.  Is there any other scripts that will resend the vm notifications in the current vm app?
12:45.16leifmadsenariel_: you can use both at the same time
12:45.30leifmadsenI'm just using MinivmMWI for MWI, and nothing else
12:45.40ariel_any sample
12:45.46leifmadsensorry, nothing that I can give out right now
12:45.53leifmadsenarticle in the future, but not for a while
12:46.04*** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk)
12:46.27ariel_ok is there a direct command that will send the notice to all that have vm?  like I set it up via a cron job?
12:47.28[TK]D-Fenderariel_: I'd go read the instructions for the app leifmadsen just handed you....
12:49.17*** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com)
12:49.20wonderworldi am having a problem with transfering calls. i assigned *1 and *2 in features.conf to enable people to transfer calls. it works, but not in a reliable way. every 5th try or so fails. they hear "transfer" they enter the extension but the call is never transfered. any idea how to debug that?
12:50.00wonderworldpeople are using SNOM 300 SIP phones
12:50.44ariel_[TK]D-Fender: t/y doing that already.  Just trying to skip heaving reading right now.
12:50.49[TK]D-Fenderwonderworld: Don't Snom's have a *real* transfer feature?
12:51.16[TK]D-Fenderariel_: "module unload lazaybastard.so" :p
12:51.26wonderworldyes i think they have. as i never used hardware-phones before i just did it in the way i always did.
12:51.31[TK]D-Fenderariel_: Jump at those freebies!
12:51.31wonderworldmaybe i should look into that
12:51.44*** join/#asterisk rossand (n=rossand@CPE000c413a19a3-CM00159a025ad4.cpe.net.cable.rogers.com)
12:51.51*** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca)
12:51.51ariel_[TK]D-Fender: yes sir,  do that every day if I can.....;0
12:51.58[TK]D-Fenderwonderworld: DTMF call-features = suck
12:52.06*** join/#asterisk manxpower (n=EWieling@24.42.221.26)
12:52.17manxpower~answers
12:52.18infobotrumour has it, answers is Asterisk Book: http://astbook.asteriskdocs.org & SIP w/ NAT: http://www.aocomputing.net/?p=3 & Wiki:  http://www.voip-info.org & docs: /path/to/src/asterisk/doc & 1.4 .vs. 1.6.0 .vs. 1.6.1: : /path/to/src/asterisk/UPGRADE*.txt
12:52.23tm1985Is there someone here that have worked with dial to *21* in dahdi and asterisk?
12:52.29*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
12:53.11manxpowerWe are supposed to know what *21* is?
12:53.31Naikrovekmanxpower: you missed that whole conversation
12:53.39voipmonklooks into one of his Dragon Balls for an answer
12:53.40[TK]D-Fendertm1985: Thogh I've never tried that specific combo I see no reason for it to work any differently than anything else
12:53.41wonderworld[TK]D-Fender: do you think playing with inband/outofband transmission of DTMF could improve things?
12:53.54wonderworldi think i saw such a feature in the phone docs....
12:53.59[TK]D-Fenderwonderworld: No way should a SIP hardphone ever be doing inband...
12:54.11wonderworldok, i'll check what mine do
12:54.25manxpowerall SIP phones support out of band DTMF
12:54.31wonderworldprobably thats it
12:54.40wonderworlddidn't configure them at all
12:54.42tm1985So why doesn't it work then
12:54.43manxpowerYou can only send inband DTMF over ulaw/alaw so it's pretty pointless over any other codec
12:54.53voipmonkbut its fun to see a bunch of 22222's when you only press one 2 , isnt it? :)
12:55.16voipmonkor whistle dtmf
12:55.23manxpowerWell, you can SEND inband DTMF over any codec.  It just won't come out the other end as DTMF.
12:55.43[TK]D-Fendertm1985: Your telco is telling you *21* is not valid.  why don't you ask them instead?
12:56.21Naikrovekmanxpower: how about g729?  DMTF work over g729?
12:56.29[TK]D-Fender<                  Ext: 1  Cause: Invalid number format (28), class = Normal Event (1) ]
12:56.31wonderworldi think there was a movie where a guy was able to whistle dtmf. funny scene
12:56.44tm1985the wierd thing is that when I use a phone that is using asterisk I can call *21*
12:56.46manxpowerNaikrovek: Of course DTMF works over G729, but only OUT OF BAND DTMF, not inband dtmf
12:57.16[TK]D-Fendertm1985: this is a PRI, not an analog line.  Where do you get the idea that such a number exists in PRI?
12:57.33Naikrovekin-band = same means of transmission as voice?
12:57.39[TK]D-Fendertm1985: PRI's don't do analog feature codes.
12:57.40manxpowerNaikrovek: yes
12:57.42Naikrovekbecomes part of the audio stream
12:57.43Naikrovekokay
12:58.02tm1985What does it then
12:58.03manxpowerand since the G729 codec is designed to compress voice VERY well and DTMF is not voice, it will garble voice.
12:58.08wonderworldcaptain crunch style
12:58.09Naikrovekout of band = sip signalling or some other non-compressed method
12:58.15Naikrovek?
12:58.16tm1985those analog features codes?
12:58.45[TK]D-Fendertm1985: Sure looks like...
12:59.15manxpowertm1985: what SPECIFIC feature is *21* supposed to activate?
12:59.17[TK]D-Fendermanxpower: You mean garble DTMF :)
12:59.26wonderworldtm1985: what kind of pbx are you using for the phones it is working with?
12:59.29manxpower[TK]D-Fender: that too
12:59.39tm1985Our provider offers that for call redirection!!
12:59.48manxpowertm1985: not over PRI it doesn't.
13:00.04tm1985Over what then??
13:00.13manxpowertm1985: * codes are for analog lines
13:00.21[TK]D-Fendertm1985: You don't redirect PRI's....
13:00.25*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
13:00.27fiddurtm1985: How is your analog phone connected, the one that *21* works with?
13:00.44tm1985It is connected with isdn line
13:00.53[TK]D-Fendertm1985: go call your telco.  Now.
13:00.56manxpowerI see today is typical.
13:01.02manxpowerHave fun, [TK]D-Fender!
13:01.04*** join/#asterisk tjz (n=tjz@220.255.158.226)
13:01.04*** part/#asterisk manxpower (n=EWieling@24.42.221.26)
13:01.12fiddurtm1985: isdn bri then, not pri?
13:02.37wonderworldyeah it's a bit strange that the lib is called libPRI, as you need it for BRI as well. should be called libISDN or so
13:02.57wonderworldtm1985: how many phone lines do you have with your telco?
13:03.00phixhey
13:03.02tm1985whe have a ISDN line so what have to use than
13:03.35fiddurtm1985: So, now you know what to ask your provider then...
13:03.55phixWhich one should I get out of these? --> Polycom Soundpoint IP 501, POLYCOM SOUNDPOINT IP 301, Linksys SPA 942, Linksys SPA 941
13:04.39fiddurphix: My experience of linksys-phones is quite bad :P  ...and most people here tends to recommend polycom, although I really like the snom's...
13:04.57Kattystretches
13:04.58leifmadsenphix: I like the polycoms
13:05.04leifmadsenpets Katty
13:05.19*** join/#asterisk b1u3m3th (n=b1u3m3th@office.gtek.biz)
13:05.26Kattyyawns, goes in search of caffeine.
13:05.35leifmadsenwonderworld: it was called libPRI before it had BRI support :)
13:05.40leifmadsenBRI support is relatively new
13:05.45wonderworldhehe
13:05.48*** join/#asterisk Buklov (n=buklov@213.138.71.254)
13:05.52leifmadsenKatty: I think I will make a double espresso
13:06.24Kattyleifmadsen: make one for me with ice, lowfat milk, whipped cream, 2 T of pumpkin puree, and pumpkin pie spice on top.
13:06.35leifmadsenwow... that's crazy
13:06.36kaldemarphix: IP301 is discontinued, consider 330 or 320 instead
13:06.36Kattyoh! and cinnamon
13:06.38leifmadsenI just drink mine black :)
13:06.50Kattypumpkin pie latte is amazing.
13:07.20wonderworldi drink 60% milk 40% coffee. nothing else.
13:07.32wonderworldfull fat milk of course :)
13:07.38Naikroveki also recommend polycoms
13:07.40Kattyleifmadsen: http://www.thismamacooks.com/WindowsLiveWriter/j0438740.jpg
13:07.45Kattyleifmadsen: visual reference.
13:07.46Naikrovekthough i have used snom and they're nice too
13:07.58leifmadsenKatty: delish :)
13:08.08leifmadsenKatty: I don't really like pumpkin pie
13:08.17leifmadsenwhich is too bad since lots of people seem to
13:08.26Kattyphix: and fyi, a 330 and 320 are basically the same, except the 330 has an extra network port in the back, if you need to 'line out' to another network device like a laptop.
13:08.31[TK]D-Fenderphix: forget the 30X/50X unless you have a killer deal, and then only conisder the 50X
13:08.33phixfiddur: I can't get snoms here
13:08.57Kattyleifmadsen: that's okay.
13:09.00Kattyleifmadsen: more for me ;)
13:09.02leifmadsen:D
13:09.04phix[TK]D-Fender: 501 == 110AU atm
13:09.09Kattythey make mocha lattes
13:09.15Kattywhite vanilla lattes.
13:09.17leifmadsengoes to make this caffienated beverage
13:09.20wonderworldphix: ehy not?
13:09.23wonderworldwhy
13:09.32leifmadsenget the 550!
13:09.43[TK]D-Fenderphix: how does that compare to the other models?
13:09.46Kattyget the 650 ;P
13:10.01Kattypats her poor little 501
13:10.15leifmadsenKatty: ya, I have a 501 too
13:10.16Kattyi am the keeper of relics. there are still some 500s in this building.
13:10.18leifmadsenit doesn't have G.722 :(
13:10.22Katty:<
13:10.24leifmadsenI have a 7960!
13:10.29Kattycaffeinated beverage
13:10.30Kattyshoo
13:10.30phix[TK]D-Fender: 41 AUD == SPA941, atm, (ebay so it could increase)
13:10.39Kattyshoos leifmadsen off irc.
13:10.44Naikroveki want to obtain a cisco phone to see how they're configured, but i love my polycom 320
13:10.58KattyNaikrovek: nice word. obtain.
13:11.10Naikrovekhehe
13:11.30[TK]D-Fenderphix: 941 is an OK phone...
13:11.35Kattyhow does chex get soggy so quickly?
13:11.39Kattythis should be against the law.
13:11.45NaikrovekKatty: lol
13:11.54Naikrovekyou have lowfat milk or something
13:12.20Kattydon't care for whole milk. feels like i'm drinkin a shake or somethin
13:12.26Naikrovek2% all the way
13:12.30Naikrovekyeah no whole here either
13:12.31jayteemornin Katty
13:12.35Kattymorning jaytee (=
13:12.37Kattyhugs jaytee
13:12.38Naikrovek1% also tolerable
13:12.42Kattyfat free is awful.
13:12.44jayteehugs Katty
13:12.44Kattycan't drink it.
13:12.47Naikroveksame
13:12.53Kattysame with cream cheese, and regular cheese.
13:12.56Kattyit's just... eww.
13:13.01wonderworldyou can buy fat-free milk?
13:13.04[TK]D-Fenderphix: I would not say that the 501 is worth double the 941...
13:13.12Kattyyou can buy fat free anything i think.
13:13.16phixsnom300
13:13.16[TK]D-Fenderphix: for average use
13:13.19wonderworldnot here
13:13.20[TK]D-Fenderphix: EW!
13:13.25phix[TK]D-Fender: that is bad?
13:13.25Kattywonderworld: where's 'here'?
13:13.28wonderworldwow...thats strange. fat free milk
13:13.31[TK]D-FenderSnom 300 = puny wastre
13:13.34[TK]D-Fenderwaste*
13:13.38wonderworldgermany
13:13.42Kattyah. right.
13:13.50Naikrovekphix: i swear to you the polycom 321/331 is superior
13:13.54wonderworldprobably it's available but i never saw it
13:14.14Naikrovekphix: and cheaper
13:14.32phixwht about snom m3?
13:14.47Kattywonderworld: we also have Fat Free sour cream, cream cheese, and  yogurts.
13:15.02casnikreally basic question and ya'll will probably just tell me to go read the TfoT (again) but .....    I am trying to just get a softphone to connect to asterisk and register to Asterisk. What all should I pay attention to in configs?
13:15.06Kattywonderworld: but then people think they can eat more of it, cause it's much lower in calories.
13:15.11phixNaikrovek: hmmm
13:15.14Kattywonderworld: and it doesn't really work that way.
13:15.24phixNaikrovek: where can I buy though from in AU? :)
13:15.33Naikrovekphix: good question
13:15.40Kattywonderworld: but instead it just gets them used to eating a larger portion
13:15.44phix16.50 for snom300 :)
13:15.49casniknot worried about external calls or anything because none of that is set up
13:15.54Naikrovekphix: let me do some poking around and i'll see
13:16.02Kattywonderworld: do you have any german recipes? (=
13:16.07phixsnom 320 $31
13:16.11phixNaikrovek: :D
13:16.16wonderworldtry http://www.chefkoch.de
13:16.22wonderworldlargest german cooking-site
13:16.31wonderworldi suck in cooking
13:16.33wonderworld;)
13:16.33Kattyoh i don't want a german cooking site.
13:16.36Kattyi want family recipes!
13:16.51Kattybesides, i only speak english :/
13:16.54Kattytranslating is difficult.
13:17.03Kattywhich is my own fault.
13:17.28[TK]D-Fender~101
13:17.29infobotit has been said that 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony.  You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf
13:17.35[TK]D-Fendertm1985: ^^^
13:17.43[TK]D-Fendertm1985: and go Google it.
13:18.00Kattythere's also a movie on the history channel about telephony
13:18.06Kattywhich is kinda neat to watch.
13:18.11[TK]D-Fendertm1985: BRI is "residential" grade telephony and PRI is meant for larger deployments with multiple DID's, etc
13:19.30Kattyi never knew a bri was considered residential grade.
13:19.36Kattydoes that mean it's available for residential?
13:19.47*** join/#asterisk gego (n=rick@b238085.customer.hansenet.de)
13:19.51Naikrovekphix: omg shipping to australia is insane
13:20.22*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:20.26Naikroveki used to live in sydney, plugging my former glebe address into here is making my mind overheat
13:20.35coppicethe majority of BRI deployment is for small businesses
13:20.42*** join/#asterisk x86 (n=x86@p3m/member/x86)
13:21.43Kattycoppice: :<
13:22.03Kattyboo
13:22.08wonderworldwhat is the price for a T1 in the US?
13:22.18Kattyi think we pay around 300ish
13:22.24Kattybut i could be wrong. i don't look at the bills.
13:23.35wonderworldany flat rate included or just for the line?
13:24.19[TK]D-Fenderwonderworld: Depends where and with whom
13:24.39[TK]D-Fenderwonderworld: watch that jump well over $1000 in a lot of places
13:24.45wonderworldok, because 300 seems to be pretty expensive
13:25.09[TK]D-Fenderwonderworld: $1000 should seem astronomical then :)
13:25.14[TK]D-Fenderwonderworld: And compared to what?
13:25.14wonderworldyes it is
13:25.20wonderworldwe pay EUR 99 / month
13:25.30[TK]D-Fenderwonderworld: For a full 30 channel PRI?
13:25.34wonderworldyes
13:25.38*** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq)
13:25.43[TK]D-Fenderwonderworld: What other charges?
13:26.03wonderworldthe calls are charged seperately
13:26.08wonderworld99 is for the line
13:26.39coppiceif you have a call centre (i.e. nothing but incoming calls) is it really 99 per month?
13:27.18wonderworldyes i think so. i was planing on getting a line in my house to host some dial-in services to make a little extra cash
13:27.59wonderworldmaybe they have a minimum call limit....never looked into that
13:29.12voipmonkminimum amount of simultaneous calls...
13:29.18voipmonkim sure
13:29.40voipmonkyou wouldnt go on tv american idol style and ask a few million viewers to dial your did
13:30.05*** join/#asterisk ramindia (n=balajibh@96-10.southernonline.net)
13:31.12wonderworldwell but 99 EUROS just for providing a hardly used line mustn't be such a bad business for the telco...
13:31.16*** join/#asterisk naif (n=naif@93-35-49-25.ip53.fastwebnet.it)
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13:31.27naifhi all
13:32.00ramindiaany one here success of Audiocodec MP-118 single box with multigateway asterisk register
13:34.01retentiveboyI'm getting "acl.c:376 ast_get_ip_or_srv: Unable to lookup 'dynamic'" in my logs after starting to use users.conf to setup SIP stations.  I've got "users=dynamic" in there so the phones can register along with hassip=yes, registersip=yes, and type=peer among others.  Is there some other combination that would fix this?
13:35.56*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
13:35.56*** mode/#asterisk [+o putnopvut] by ChanServ
13:37.44naifHi, i have my uplink carrier that sometimes make me some nasty joke, like giving me an "Answer" on SIP but then i get in the audio flow (while paying) a never ending ringing. There is some easy method to detect it? I read about callprogress=yes and busydetect=yes but is for PSTN lines (http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf). I need to detect this "fake ringing" after answering in a SIP call. Any idea?
13:38.06*** join/#asterisk Naikrovek (n=jjohnson@63-252-251-77.ip.mcleodusa.net)
13:38.28[TK]D-Fendernaif: No.  You are quite screwed
13:38.31Naikrovekis pissed off. again.
13:39.01naifDamn, it's something like a fraud. Cheap traffic have this drawback
13:39.09[TK]D-Fenderretentiveboy: that should be host=dynamic
13:39.12naifi pay for a call that will never be established
13:39.27KattyNaikrovek: whatsamatter
13:40.00retentiveboy[TK]D-Fender: ah, been starting at it too long.  much thx
13:40.34Naikrovekthis is 2009.  have we not gotten far enough along to create some sort of corporate friendly downloading application, let's call it a "downloader" that can reliably resume from interrupted transfers?
13:41.01Naikroveki love how one guy wrote bittorrent and yet all these businesses have failed to come up with something better
13:41.11Naikroveki'm talkin' to you, microsoft
13:41.20KattyNaikrovek: ah. well.
13:41.21retentiveboy[TK]D-Fender: wait, typo on my part.  I do have host=dynamic.
13:41.25Naikrovektheir download manager is absolute garbage
13:41.26KattyNaikrovek: not much you can do about that.
13:41.32KattyNaikrovek: so perk up buttercup!
13:41.49Naikrovekit can't throttle, it can't schedule, it can't reliably resume
13:41.54Naikrovekbittorrent can do all of that
13:42.00Naikrovekand ONE GUY designed it
13:42.20ramindia[TK]D-Fender:  hi
13:42.29Naikrovekso, rant is over
13:42.36Naikrovekhappy face
13:42.51Kattyyay!
13:42.54[TK]D-Fenderretentiveboy: that's what I told you...
13:43.12Naikrovekputs the headphones on, connects to 24/7 Loveline stream, and mellows out.
13:43.57*** join/#asterisk oej (n=olle@132.177.253.250)
13:44.01retentiveboy[TK]D-Fender: yeah, I mistyped my question.  I have host=dynamic, hassip=yes, registersip=yes and type=peer in there and am getting that error.  Sorry for the confusion.
13:44.21Naikroveki need to integrate cacti with asterisk. hrm.
13:44.23Naikrovekgoogles..
13:44.36[TK]D-Fenderretentiveboy: stations should not have "registersip"
13:44.49*** join/#asterisk xpot (n=james@70-91-210-233-BusName-Utah.hfc.comcastbusiness.net)
13:45.08[TK]D-Fenderretentiveboy: by the time you're done pastebin its config & "sip show peer [thepeer]"
13:45.10[TK]D-Fender~pb
13:45.11infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
13:45.11[TK]D-Fender^^^^^^^^
13:46.19leifmadsenNaikrovek: there is a how-to for that. Hold on, I'll find the link I saved.
13:46.25Naikrovekooh nice
13:46.32Naikrovekthanks, leifmadsen
13:46.37Kattycacti?
13:46.49Kattypictures big spikey green plants with red leaves.
13:46.57retentiveboy[TK]D-Fender:  will do if removing that doesn't fix it.  Thanks.  btw, I've been in the code looking for settings that various modules are looking for in users.conf.  Should the users.conf sample be updated to include some of these?
13:47.12Naikrovekcacti = pseudo monitoring.  i can monitor and graph bandwidth usage, cpu usage for systems, etc
13:47.36Naikrovekit doesn't alert if something is unreachable or passes a limit or whatever, just collects info and graphs it
13:47.56retentiveboy[TK]D-Fender: error's not coming up now.  thx
13:50.39retentiveboyNaikrovek: tried enabling the SNMP agent in * and polling that from Cacti?
13:50.51Naikrovekhowcome TI graphing calculators cost $140, when Casio graphing calcs, that are programmable and do everything a TI calc can do, cost $40
13:50.57Naikrovekretentiveboy: yeah working on that now
13:51.40retentiveboyNaikrovek: I'm curious what you can monitor from there...
13:51.47Naikrovekwe'll see
13:51.53coppiceNaikrovek: schools require TI calculators
13:52.00[TK]D-FenderNaikrovek: Because TI is in collusion with major schools to force students to buy them for their classes
13:52.03leifmadsenNaikrovek: well, apparently I didn't bookmark it, and I can't seem to find it
13:52.13*** join/#asterisk ehsjoar (n=ehsjoar@c-24-9-91-203.hsd1.co.comcast.net)
13:52.17Naikrovekleifmadsen: i found this: http://www.voipphreak.ca/2007/04/16/monitoring-asterisk-14-with-snmp-and-cacti-for-pretty-graphs/
13:52.22[TK]D-FenderNaikrovek: ca$h c0w
13:52.28leifmadsenNaikrovek: ya, there was a newer one actually that I saw the other day
13:52.51Naikrovek[TK]D-Fender: i guess.  the components that the calc is made of cost maybe $35 total
13:53.02mutantecould you help me with this: I have Asterisk configured as client to a SIPgate.de as provider, i can make calls to external phones, just i cant "Playback" a message. this is how i try right now  http://pastebin.ca/1568311
13:53.58casnikso if I set 2 [Xlite] phones in sip.conf , does that mean they "register" with the entensions I put in there for them?
13:54.16ramindia[TK]D-Fender:  what is the best tool to use.. to identify voice breaks..choppy voice..how can i identify this..
13:54.20casnikare they suppose to just be able to call each other
13:54.20retentiveboyNaikrovek: doc/asterisk-mib.txt
13:54.29Naikrovekretentiveboy: thank you
13:55.06coppiceseems a strange world where a programmable calculator runs Linux
13:55.40retentiveboyNaikrovek: I build my * machines with the SNMP agent enabled hoping to get time to hook them up to Cacti.  Looks like I'm going to have to get busy :)
13:56.00Naikrovekretentiveboy: how many * boxes do you have
13:56.07[TK]D-FenderNaikrovek: Rackets... not just for tennis anymore ;)
13:56.12retentiveboyNaikrovek: 6
13:56.15Naikrovek[TK]D-Fender: no kidding
13:56.18casnikI have two instances of Xlite running on two desktops , they are getting the congratulations auto greeting from asterisk ... but I am not getting them to call each other >.>
13:56.28[TK]D-Fenderramindia: Don't know, and please avoid targeting individuals for questions like this
13:56.47Naikrovekleifmadsen: this one? http://www.voipphreak.ca/2008/10/28/asterisk-snmp-with-cacti-howto-upgraded-for-asterisk-16-and-ubuntu/
13:56.54ramindia[TK]D-Fender:  got you
13:56.55leifmadsenNaikrovek: ah, that might have been it!
13:56.56leifmadsen:)
13:56.59[TK]D-Fendercasnik: Good odds they are fighting for your SIP port on that machine.
13:57.19[TK]D-Fendercasnik: You'll have to run them on separate ports, and configure it to match in their peers
13:57.51casnik[TK]D-Fender, ok I'll try to figure out how to do that next then
13:57.53casnikty
13:58.02wonderworldmutante: Your announcemen is never played, because "Dial" ends, when the call is finished and one site has hung up
13:58.29wonderworldmutante: you are looking for the "A"-option of the Dial() command
13:58.29casnik[TK]D-Fender, in the sip.conf right?
13:58.55Naikroveki can't believe how stupidly complex this is.  i so hate linux.  as linux things go, this isn't complex at all, as real software things go, this is unacceptable
13:59.00[TK]D-Fendercasnik: yes
13:59.22casnik[TK]D-Fender, cool
14:00.39*** join/#asterisk wathek (n=wathek@41.224.194.132)
14:01.45*** join/#asterisk deeperror (n=deeperro@adsl-76-226-149-104.dsl.sfldmi.sbcglobal.net)
14:02.28casnik[TK]D-Fender, not seeing it where I set up locally connected SIP devices ... no mention of port ... amidoinitrite?
14:03.12[TK]D-Fendercasnik: its up to you to put it in there
14:03.19MarcWeberCan I make Skype calls from asterisk?
14:03.27[TK]D-Fender~skypeforasterisk
14:03.28infobotrumour has it, skypeforasterisk is a Digium-made channel driver that allows Asterisk to connect to the Skype network using a Skype-supported connection method. Unlike all other solutions it does not require X11, kernel modules, or a Windows machine. See http://www.digium.com/skype for details
14:03.31[TK]D-Fender^^^^
14:03.32casnikso just like port=12000
14:03.48[TK]D-Fendercasnik: I'd recommend one on 5060, the other on 5061 for instance
14:04.24casnikreally .... so I can just count up from there
14:05.07*** join/#asterisk retentiveboy (n=pdugas@atl.pra-corp.com)
14:06.08casnikbah , still gave me call failed , ((( chan_sip.c:14721 handle_request_invite: Call from '' to extension '1001' rejected because extension not found. ))
14:06.36[TK]D-Fendercasnik: that is a DIALPLAN error
14:06.41casnikyeah
14:06.43*** join/#asterisk andres833 (n=andres83@190.144.75.22)
14:06.44[TK]D-Fendercasnik: extensions.conf <-
14:06.50casniktime to do all that
14:06.53MarcWeber[TK]D-Fender: Thank you!
14:06.55*** join/#asterisk moy (n=moy@mail.e-contact.cl)
14:07.02casnikat least I read that chapter lol
14:07.07*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
14:07.20[TK]D-Fendercasnik: You'd better master the dialplan, because that is 95% of Asterisk
14:07.27casnikyeah
14:07.37Kattydecides on stroganoff for lunch.
14:07.45casnikjust wanted to get to where I could connect a phone ... then was gonna go to that
14:10.15*** join/#asterisk wcselby (n=wcselby@216.110.88.254)
14:12.15*** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net)
14:12.48Naikrovekneeds to master the dialplan
14:14.43casnikneeds to L2 dialplan.
14:16.35mutantewonderworld: thank you, i saw the options to the Dial command in the console, but i have no idea how to write it in a call file
14:17.50wcselbybmoraca - i was able to successfully get two softphones (on two separate computers) to register to my asterisk server last night, through my 2wire
14:18.14wcselbybmoraca - I use at&t u-verse internet
14:21.15Naikrovekwcselby: nice
14:21.41*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
14:21.43wcselbyNaikrovek - he asked me yesterday if I could get two phones connected through a 2wire, I told him I'd check.  So I did.... :)
14:22.40mutanteso if i have  "exten => _X.,2,Dial(SIP/${EXTEN}@sipgate-out,30,trg)"  now, where would i put the "A" option to Dial()
14:23.11wcselbymutante - with the 'trg' part
14:23.23wcselbymutante - so it would become ,trgA)
14:23.35wcselbybut I don't think that's the proper spot for options in the Dial() command, let me check
14:23.38mutantewcselby: ahaa, and the name of the sound to play?
14:24.32wcselbymutante - you would enter ,Dial(SIP/${EXTEN}@sipgate-out,30,trgA(nameoffiletoplay))
14:24.33wonderworldexten => _X.,2,Dial(SIP/${EXTEN}@sipgate-out,30,trgA(mysoundfile))
14:24.42mutantethank you :)
14:25.12wonderworld"mysoundfile" mustn't have a file extension
14:25.21wonderworldso it's not mysoundfile.wav
14:25.25mutanteyep, i learned that yesterday :) ...trying
14:25.27mutantei have .gsm files
14:25.27wonderworldjust mysoundfile
14:26.31mutantearr, nope, still hangs up..:( this would have been to easy...cru
14:26.56wonderworldcheck the CLI to see what is happening....
14:27.23wcselbydid you do a dialplan reload after you made the change to the file?
14:27.25wonderworldthe A option works for sure, i am using it a lot
14:27.34mutante<PROTECTED>
14:27.36wcselbyi forget that every now and then when doing lots of small changes
14:27.56mutantewcselby: i did a /etc/init.d/asterisk restart
14:28.00wonderworldthere seems to be an extension missing....
14:28.16wonderworldor did you remove it?
14:28.32mutante?
14:28.41wonderworldSIP/.....   did you remove the number?
14:28.44mutanteyes
14:28.47wonderworldahh ok
14:29.03wcselbymutante - paste the whole output from the cli for the call to a pastebin
14:29.04mutantethe phone on my desk rings
14:29.11mutanteit just doesnt play the message after pickup
14:31.05wonderworldincrease the verbosity on the cli and post the full call output to a pastebin
14:31.52*** join/#asterisk xrmx__ (n=rm@host103-251-dynamic.15-87-r.retail.telecomitalia.it)
14:32.40*** join/#asterisk csmyth (n=csmyth@ext-52.sagetelecom.net)
14:34.50mutantehttp://pastebin.ca/1568366
14:34.53*** join/#asterisk Skeeter- (n=wil_c_wi@190-141.cgocable.ca)
14:35.09*** join/#asterisk user4545 (n=sipip@p57B1F38A.dip.t-dialin.net)
14:36.45[TK]D-Fendermutante: [Sep 16 16:28:05] WARNING[1625]: pbx.c:3080 pbx_extension_helper: No application 'SetCallerId,SIPID' for extension (sipout, 10, 1)
14:37.09mutanteso if i dont set a caller id i can still call, but not play sounds?
14:37.27[TK]D-Fendermutante: I don't see you showing us your configs or the call file...
14:37.30*** join/#asterisk hugorebelo (n=hugorebe@200-171-132-124.completo.com.br)
14:37.58user4545Hi, my dialplan MENU work nicht...I call one number in Asterisk from outside and i cann't go to next menu, Asterisk don't understand my additions digits
14:38.12user4545anybody can help me?
14:38.21Naikroveksettle down
14:38.29Naikrovekdon't expect a quick response
14:38.35Naikrovekjust wait and see who answers
14:38.47user4545<PROTECTED>
14:38.47user4545[Sep 16 16:33:52] WARNING[17073]: pbx.c:5656 pbx_builtin_waitexten: Timeout but no rule 't' in context 'von-voip-provider'
14:38.47user4545<PROTECTED>
14:39.04wcselbyuser4545 - pastebin your extensions.conf file (or at least the relevant parts) and give us a link
14:39.13wcselby~pb
14:39.14infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
14:39.14user4545one moment
14:39.15*** join/#asterisk kondela (i=kondela@116.68.103.250)
14:39.22[TK]D-Fenderuser4545: You took too long to respond and * terminated the call because you didn't have a "t" exten to handle the fact they took too long
14:39.47Naikrovekthe DMTF could have been garbled or lost via the voice encoding as well
14:39.59wcselbycould also be a digit timeout issue
14:40.01*** join/#asterisk wopsy (n=80475@AToulouse-754-1-13-64.w90-55.abo.wanadoo.fr)
14:40.11wcselbyif he's trying to dial a two or three digit number
14:40.32Naikroveks/DMTF/DTMF/
14:40.47user4545please http://pastebin.com/m74b7e501
14:40.52kondelahi i do have a queue related question
14:40.57*** join/#asterisk fenlander (n=fenlande@82.152.81.57)
14:41.26user4545how van I DMTF ON-make ?
14:41.27mutante[TK]D-Fender: http://pastebin.ca/1568373
14:41.30user4545howcan
14:41.53*** join/#asterisk ruben23 (n=RPL@122.55.48.243)
14:42.43[TK]D-Fendermutante: "core show function CALLERID" <- and go fix your dialplan
14:42.43wcselbyuser4545 - are you using a compressed codec?
14:43.00[TK]D-Fendermutante:  it is dying on that illegal app name
14:43.04user4545wcselby: no I have just asterisk instaled
14:43.05kondelaif i use agentlogin(), i cant see the callerid of the caller, because the agent logged in the queue..
14:44.21mutante[TK]D-Fender: well ok, how come it still dials though...hmmm
14:44.28*** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net)
14:45.18kondelahi..  can anyone share with my thoughts..?
14:45.34wonderworldmutante.... please post the output of a call.... before that: "core set verbose 100" "sip set debug off"
14:45.52wcselbymutante - exten => _X.,1,SetCallerId,SIPID - should be exten => _X.,1,Set(CALLERID=something)
14:46.19mutanteok, grr, that line is the one i got as example from the SIP provider page itself :P
14:47.03wonderworldyes, you have to replace "SIPID" with your actual SIP-ID. the one you got from your provider
14:47.06wcselbymutante - I think maybe they were giving you guidelines or something
14:47.20mutantei see, thanks
14:47.21wcselbyas opposed to something you copied and paste
14:48.27wonderworldmutante: can you talk on the phone when you are called?
14:48.35wonderworldor is the call directly hung up?
14:48.41*** join/#asterisk kerchunk (n=kerchunk@pool-173-49-10-152.phlapa.fios.verizon.net)
14:48.51mutanteit rings and after pickups is directly hung up
14:48.52garymcanyone in here got a spare minute to test my sip extension?
14:49.12garymcit was working earlier, but some guy i test with now doesnt hear nothing. Unless its his sound card?
14:49.58mutantewonderworld: http://pastebin.ca/1568382
14:49.59wcselbygarymc - i can make a call to you if that's what you're asking?
14:50.06wcselbyPM me with a number to call
14:50.09*** join/#asterisk heit0050 (n=heit0050@mail2.heitkeconsulting.com)
14:50.36kondelai do have a general regarding queue behaviour in asterisk..  can someone assist me..
14:50.45*** join/#asterisk Tim_Toady (n=moi@adsl194-8.kln.forthnet.gr)
14:52.13wonderworldmutante: the call file is calling you at 0211-something and another side at sipgate-out/10
14:52.27wonderworldare you sure sipgate-out/10 is a valid number with your provider?
14:53.14mutanteno, i am not, can i just remove the "Extension: 10" from the call file
14:53.47wonderworldwhat do you want to do with your call-file?
14:53.55wonderworldwhat triggers it?
14:54.03mutantei want to call 0211-something and play a message
14:54.28kondelahello world..  how can i display a caller id , while the agent is logged in using agentlogin()
14:54.29*** join/#asterisk |Cybex| (n=John@80.100.126.176)
14:54.32wonderworldok
14:54.37mutanteafter this works we will probably make nagios move a callfile to "outgoing" in the case of an alarm
14:55.55wonderworldok, your extensions.conf is not correct for that setting
14:56.05*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
14:56.07wonderworldyour callfile calls 0221-something, which is your desk, i asume?
14:56.19mutanteright, and it does ring
14:56.25wonderworldafter that, it tries to connect you to sipgate-out/10
14:56.44wonderworldbecause you said in the call-file, it sould go to extension 10 after connecting with 0221-something
14:57.25wonderworldjust create an extension called "10" and put a playback in there
14:57.37wonderworldyou don't need to dial to any other party
14:57.42mutanteaha, ok, but "At least one of app or extension must be specified"
14:57.48mutanteah, ok
14:59.18*** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net)
14:59.42wonderworldit went into your _X. extension, becuase thats a wildcard for "any number"
14:59.51*** join/#asterisk outtolunc (n=me@c-98-248-96-110.hsd1.ca.comcast.net)
14:59.52wonderworldit matches "10" as well
15:00.02kondelahello...  any  queue expert here..:)
15:00.05wcselbykondela - have you tried messing with Set(CALLERID) ?
15:00.45kondelawcselby.. finally.. thanks.
15:01.07kondelaso you are saying i have to set a caller explicitly..?
15:01.32*** join/#asterisk DigitalFlux-AFK (n=DigitalF@unaffiliated/digitalflux)
15:01.52DigitalFlux-AFKHey everybody
15:02.11DigitalFlux-AFKI need some help regarding how queues are handled in Asterisk ..
15:03.16wcselbykondela - I'm not sure, I don't really know what's going on with your situation.
15:03.36kondelawcselby - let me put it straight
15:04.08[TK]D-Fenderkondela: Why start now?  I mean you've been in here 20 minutes already and never asked a real question...
15:04.26kondelawell i asked..  may be you didnt see
15:04.48[TK]D-Fenderkondela: Answer : you CAN'T
15:05.15[TK]D-Fenderkondela: Create an external app that monitor AMI to trap the AgentConnect message
15:05.30[TK]D-Fenderkondela: This will require a PC side app
15:05.48*** join/#asterisk plundra (i=404@article.se)
15:05.54kondelaAAHA..
15:06.01kondelainteresting
15:06.28kondelaso the answer is with agentlogin(), an agent cannot see the callerid
15:07.31*** join/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-23-22.w90-56.abo.wanadoo.fr)
15:07.41plundraOk, this might be a common problem... I've got a few SPA942 setup, calling out via my asterisk is fine, but when calling each other, the mediastream seems to get lost. No sound what so ever. NAT is not used anywhere and all clients have external addresses.
15:07.51kondela[TK] so the answer is with agentlogin(), an agent cannot see the callerid..
15:08.18[TK]D-Fender[11:04]<[TK]D-Fender>kondela: Answer : you CAN'T <- was this somehow not clear?  How much more clarification do require?
15:08.32merlin8282Hi all ! How can I access data a script sets (such as environment var, or a file, or anything) to work with it in asterisk ?
15:08.36kondela[TK] though i never used agentcallbacklogin(), i think this allowed us see the callerid.. a m i right..?
15:08.54*** join/#asterisk davidandgoliath (n=David@out.clearnet.com)
15:09.14kondela[TK]  your previous message is clear to me
15:09.28kondelai think i mis-typed
15:10.06mutantethank you all guys.. i got it working:)
15:10.28[TK]D-Fenderkondela: AgentLogin is YOU calling into an app.  AgentCallbackLogin is setting * to call a local channel to ring the phone.  that is a call.
15:11.16[TK]D-Fendermerlin8282: "core show function ENV" , "core show function STAT"
15:12.16kondela[TK] i got it right..  my question was can the agentcallbacklogin()  can present original callerid to the agent.  i cant try this, since i dont have any 1.2 * withme now..  do you have any experience this part
15:13.12[TK]D-Fenderkondela: its a call.  callee's get CALlERID.
15:13.21kondelaok..
15:13.44merlin8282[TK]D-Fender: I already know these functions. The proble is, I can't set the environment variable from within my script.
15:14.04[TK]D-Fendermerlin8282: Then go have it set something else.
15:15.03merlin8282I don't understand what you mean.
15:15.12kondela[TK] is there any replacement for agentcallbacklogin() in 1.6/1.4 release..  or an other method to implement the agentcallbacklogin() behaviour in those latest release..?
15:16.54*** join/#asterisk Whitor (n=Whitor@rrcs-24-97-4-146.nys.biz.rr.com)
15:17.41leifmadsenkondela: the functionality is available via dialplan
15:18.36garymcok one more, anyone else with a mic can test my asterisk pbx by logging into it with a softphone. Take 2mins of your time?
15:18.50leifmadsenkondela: http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbacklogin-to-standard-dialplan-methods-part-1/  <-- might be useful
15:18.51[TK]D-Fenderkondela: http://leifmadsen.wordpress.com/tag/addqueuemember/
15:18.56leifmadsenheh :)
15:19.18kondelaalright.. alright... thats awesome response..
15:19.20Naikrovekgarymc: i don't have a softphone, nor the desire to set one up, nor the time, but otherwise i'd be happy to help heh
15:19.34leifmadsenI still have to finish that article, ugh
15:19.36kondelai am reading this now
15:19.39leifmadsentoo much work! :)
15:19.48jayteeTRABAJO!
15:20.29garymcNaikrovec : Sarcasim?
15:20.46[TK]D-Fenderkondela: All you really need is to seriously read the instructions for AddQueueMember and RemoveQueueMember
15:21.04ruben23jaytee: Pinoy....
15:21.45kondelaaha..  thats what i found, googling while chat..
15:23.38garymcanyone?
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15:25.33rene-hey guys
15:26.05rene-can i lower my t1 costs if i order my t1s to be dropped at a carrier hotel?
15:26.35Qwellrene-: maybe.  call and ask
15:28.10Naikrovekgarymc: not sarcasm, i'm just busy and i don't dig softphones.  have your friend test his sound in some other way
15:28.21Naikroveksoftphones imho are not worth the time
15:29.47wonderworldtwinkle is nice
15:29.52[TK]D-FenderNaikrovek: No need to use a softphone to test for him
15:30.10Naikrovek[TK]D-Fender: i know but i'm just too tired to put any effort in anything right now
15:30.17[TK]D-FenderNaikrovek: He was over-specific for no valid reason.
15:30.35Naikrovek[TK]D-Fender: well it's a common thing, to do that.
15:30.39[TK]D-FenderNaikrovek: But not busy enough to simply ignore him ;)
15:30.47*** part/#asterisk merlin8282 (n=merlin82@AStrasbourg-753-1-23-22.w90-56.abo.wanadoo.fr)
15:30.49garymcwell you can use asip phone if you wish?
15:31.17Naikrovek[TK]D-Fender: i'm at this weird place where i'm too busy to take on more activity but too bored to leave it.
15:31.18[TK]D-Fendergarymc: Is that a question?
15:32.00Qwell[TK]D-Fender: no?
15:32.14garymcyou are all to clever for me
15:32.16[TK]D-FenderQwell: :p
15:32.29Naikroveksorry garymc, someone will test for you i'm sure, just be patient
15:32.58wonderworldgarymc: i would if setting up my usb-mic wouldn't be such a pain with linux
15:33.52[TK]D-FenderNaikrovek: http://www.youtube.com/watch?v=C_Y6231uAmo <- Focus on 1:23
15:34.03[TK]D-FenderNaikrovek: But completely worth the full view
15:34.06[TK]D-Fender(listen)
15:34.15user4545can me anybody help with it ? http://pastebin.com/m273224f4 .... I cann'nt callthrow.. my Asterisk don't understend DTMF digits
15:34.20Naikroveklol
15:34.25Naikrovekhow long will that take
15:37.46[TK]D-Fenderuser4545: try "dtmfmode=inband"
15:38.07user4545one moment
15:38.24[TK]D-Fenderuser4545: actually, some guides I just googled said "info" instead
15:38.54Naikroveki wish google handled regexes
15:39.23user4545no results
15:39.44user4545donn't working
15:42.53[TK]D-Fenderuser4545: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk
15:43.10[TK]D-Fenderuser4545: if it doesn't work, go contact them
15:46.28user4545it's work, but I want different number call throw DTMF
15:46.56[TK]D-Fenderuser4545: Please rephrase that...
15:47.53wonderworlduser4545: check if you are using "rfc2833" as signalling method for DTMF in your sip-phone / softphone
15:47.59*** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net)
15:49.43wonderworldthere are different ways of transmitting DTMF. checking the sipgate example conf, they seem to be expecting "rfc2833"
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16:14.59wcselbyawesome -> http://www.voip-info.org/wiki/view/Paradise+Tent+for+Camping+and+Holiday+Needs+
16:16.33Naikrovekhehe
16:16.38wonderworldthat sucks
16:16.48wcselbythere were links there
16:16.54wcselbyi removed them, obviously
16:17.27wonderworldsome guy or some bot posted that?
16:17.43wcselbyWed 16 of Sep, 2009 [09:17]dielonput125.60.173.1481
16:17.48Naikroveksomeone was using it to vent
16:17.58wcselbyno they used it to spam
16:18.05wcselbyi changed it to "No."
16:18.22wcselbythe original -> http://www.voip-info.org/wiki/page_history.php?page_id=5762&preview=1
16:19.48wonderworldhis website sucks. right sidebar says: "My right sidebar goes here. "
16:20.19wcselbyhaha
16:20.30wcselbythe left sidebar contains all the viruses he wants to install on your pc
16:20.39wcselby:P (j/k, haven't been to the site)
16:21.26wcselbywhat was the lawnmower one?
16:21.28*** join/#asterisk maour (n=gnu@unaffiliated/maour)
16:21.29wcselby~lawnmower
16:21.34wcselbyor something like that
16:21.44Naikrovekboschlawnmower?
16:21.44afinkis there a general ration of seconds/ring?
16:21.57Naikrovekseconds per ring?
16:21.59wcselbyhttp://www.voip-info.org/wiki/view/Bosch+lawnmowers
16:22.25afinklike 5 = 1 ring or so
16:22.44Naikrovekoh a general ratio
16:22.46Naikrovekyou said ration
16:22.55Naikrovekno idea
16:23.02afinkoh whoops
16:23.07Naikrovekpretty sure they're more or less the same across the US, but i have never timed it
16:23.11wcselbyi think 5 seconds to ring is about right, depends on the country you're in
16:23.18Naikroveki was going to say 3 seconds
16:23.23Naikrovekbut i have no idea if that's right
16:23.41Naikrovekin australia, their ringing sound sounds like our busy signal
16:23.46Naikrovekif i recall
16:24.02Naikroveki remember thinking that everyone was busy
16:24.02Naikrovekbut their phones were all ringing
16:24.06wcselbylol
16:24.18Naikrovekyeah so lots of people thought i was pranking them
16:25.11wonderworldi did a funny prank a few years ago.
16:25.20wonderworldconnected to pizza ordering service with one another
16:25.33wonderworldboth phones were ringing
16:25.36Qwellwonderworld: You're an ass.  heh
16:25.41Naikroveklol
16:25.51wonderworldit took them a minute to find out, that noone wanted to order
16:25.55wonderworldquiet funny
16:26.08wcselbyspeaking of phone pranks
16:26.09wonderworldnot to...two of course
16:26.20wcselbydid anyone read leifmadsen's blog post about telemarter torture?
16:26.25*** join/#asterisk garymc (n=garymc@host81-134-0-102.in-addr.btopenworld.com)
16:26.52wonderworldQwell.... i was young
16:27.56wonderworlddid the same thing with a nazi and a communists party office. funny too
16:29.40Qwellwonderworld: hope you don't ever need to get a security clearance..
16:31.02*** join/#asterisk Tim_Toady (n=moi@adsl194-8.kln.forthnet.gr)
16:31.02wonderworldnah, don't want one. if they have to go thru my call history of the last 10 years for it, i want it even less....
16:31.59coppiceyou can always get one at a clearance sale
16:32.12wcselbycoppice.................
16:32.15wcselbythat was bad
16:33.51leifmadsenwcselby: I did :)
16:34.33Qwellleifmadsen: why would you read that nubs post?
16:34.38Naikrovekhah
16:34.49wcselbyhaha @ leifmadsen
16:34.50wcselby:)
16:35.17leifmadsenQwell: BURN
16:36.23russellbleifmadsen paid off the other 2 authors to get his name on the book ...
16:36.34leifmadsenthat's why it's at the end of the list
16:36.50QwellI bribed leifmadsen to get a mention in the book.
16:36.52Qwelltotally did
16:36.57leifmadsenit's true
16:37.26Qwell(that stills cracks me up btw)
16:37.53Naikrovek... how uh.. how much did that cost
16:38.03QwellNaikrovek: about a beer
16:38.09Naikrovekhrm.
16:39.33wcselbylol, where's the mention?
16:39.34Naikrovekso a case would get maybe contributor credit then?
16:39.37wcselbynow I'm curious
16:39.48wcselbyNaikrovek - depends on the type of beer I would imagine
16:39.50Naikrovekwe'll have to know qwell's real name
16:39.57Qwellnot like it's hard to find
16:40.42wcselbyjust got an email from my helpdesk that our internet connection was down
16:40.45wcselby....
16:40.55*** join/#asterisk DrCarumas (n=Carumas@adslfixo-b3-127-186.telepac.pt)
16:40.58DrCarumasHi!
16:41.08Naikroveki love how people email me saying email is down
16:41.13Naikrovekhappens nearly weekly
16:41.40wonderworldexchange server, huh?
16:41.48wcselbywonderworld - hahahahaha
16:41.54wcselbyi was thinking the same thing
16:42.05Naikrovekyes, but email has never gone down without the power going down
16:42.34wonderworldand power is going down once per week?
16:42.37Naikrovekthey're just dumb
16:42.46wcselbyi used to get calls from people saying "my email doesn't work".  I tell them to try sending me a test.  I'd get it.  "Well, you're email isn't down.  So what's your specific problem".  "Oh, I got an error trying to send so and so an email"
16:42.49Naikrovekno, power rarely goes out (once or twice in previous 6 months)
16:43.05wcselby"that doesn't mean email is down"
16:43.15Naikrovekyeah that's similar to what i get
16:43.38DrCarumasguys, i'm using a sip provider to place inbound/outbound calls. Sometimes oubound calls wont work for a while and i get this error: Got SIP response 503 "Service Unavailable gkd" back from "ipaddress" . I've google it but not much information. Do you think this could be something with my asterisk (v.1.4.24.1) or is from my voip provider? Thanks in advanced.
16:44.00wcselbythen, since I'm not a BoFH, I'd help them fix their problem and everyone would be happy..........
16:44.02NaikrovekDrCarumas: do all outbound calls fail or just some
16:44.07wcselbyhonest, that's what happened
16:44.14DrCarumasNaikrovek, wen this appens all fail
16:44.35DrCarumasNaikrovek, same error, then after a while everything is back ok.
16:44.36NaikrovekDrCarumas: do you check with your provider when this happens?  where does the SIP 503 come from
16:44.39wcselbyDrCarumas - I'd say you need a secondary, failover provider
16:45.05Naikrovekyeah this reeks of provider issues
16:45.23DrCarumasNaikrovek, i've openned a support ticket i'm still waiting but wanted your opinion
16:45.35DrCarumaswcselby, that's true i realy should
16:45.41wcselbyDrCarumas - I'd say service provider issue.
16:45.57NaikrovekDrCarumas: my provider does this, well their upstream provider does this, so once in a while, phone calls to a specific area will fail to go through
16:46.01DrCarumaswcselby, ok i'll wait to see what the ticket resoltin will conclude.
16:46.14wcselbylol, someone behind me just said "uh oh, is Microsoft down too?"
16:46.16NaikrovekDrCarumas: this is almost certainly a provider issue, or their network provider
16:46.43DrCarumasNaikrovek, ok. Thanks for your help and wcselby.
16:46.50wcselbyDrCarumas - np
16:47.32*** join/#asterisk Carlos_PHX (n=carlos@68.108.193.174)
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16:55.26ZenBSDiexten => ${ZenBSDi},1,Background("I got da skillz!")
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16:56.14wcselbyZenBSDi - drop the quotes
16:56.21wcselby:)
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16:59.49ZenBSDiexten => ${wcselby},1,Gotoif($["${ZenBSDi}" = "don't care"]?hatepeoplewhokillmyfnjokes,11
17:00.03QwellThat was a joke?
17:00.56ZenBSDiof course.. cause I got no skill :p
17:01.04wcselbyZenBSDi - lol
17:01.28wcselbyhey, if it was [TK]D-Fender responding, he wouldn't have been as nice as I was :P
17:02.01ZenBSDiYeah and I'd move him to the hangup extension too :p
17:02.29ZenBSDiOr worst.. run him through an AGI script that dices him up in /dev/null :p
17:02.42*** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
17:02.42[TK]D-FenderZenBSDi: O RLY?
17:02.54wcselbylo
17:02.58wcselbylol even
17:03.09casniktheres a storm comin!
17:03.19ZenBSDi:p
17:06.06ZenBSDiWho AGI scripts in here and who prefers perl or php for agi programming?
17:06.26[TK]D-FenderZenBSDi: Yes
17:06.27casnikIf I was at that point I would pcik perl
17:06.34casnikpick*
17:06.39casnikbut I am still newb
17:08.16Qwellbah.  name one thing you can do in AGI that you can't in pure dialplan
17:08.43ZenBSDiI wrote a small phpagi script to do credit card processing ... just sucks having to use the weak text to speech stuff thats out there.
17:09.32ZenBSDiOwell, Control your pbx from a web interface :p
17:10.47ZenBSDibut thats administration :p .. agi vs dialplan.. well .. I guess it comes down to dynamics. changing passwords or extensions in a database and having it agi check or agi set against those database variables maybe
17:11.36ZenBSDiI like setting up Asterisk realtime with MySQL .. so I'm odd like that
17:12.08[TK]D-FenderQwell: Perform background tasks while playing audio outside of an IVR (non-interruptable)
17:13.07ZenBSDiheh... nice
17:13.40wonderworldi did a server-remote-control thingy with AGI/php. but just to learn agi. i think it would have been more complicated within the dialplan, but maybe not....
17:14.54*** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be)
17:15.21wonderworldthe thing can read me my email, don't know if that would have been possible within the dialplan. there was a lot of text-parsing involved.
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17:22.51p3nguinIf my phone is on the same LAN as the Asterisk server, but the provider is on the outside of NAT, should the peer be configured for nat=no or nat=yes?
17:23.05p3nguinpeer being the phone
17:24.39IBC_jkenneyIs there any plans for digium to allow hints from the realtime mysql engine?
17:24.49IBC_jkenneyin any new releases of asterisk or asterisk addons
17:24.49p3nguinAnd what about for the peer context of the provider?  Their config says "; nat=yes ; Uncomment this if your box is behind a NAT"
17:25.20p3nguinThe Asterisk box is the gateway between the internet and the LAN, so is Asterisk behind NAT or not?
17:27.18Naikrovekp3nguin: i use nat=yes
17:27.26Naikrovekp3nguin: asterisk box is also router?
17:27.32p3nguincorrect
17:27.36Naikrovekew
17:27.39Naikrovekwell if it works...
17:28.01Naikrovekdepending on which interfaces * is listening on, it could be on the LAN, WAN, or both
17:28.07Naikrovekprobablyboth
17:28.15p3nguinIt listens on both.
17:28.24Naikrovekit straddles both then
17:28.33p3nguinThis provides a public IP address for Asterisk, so I'm thinking Asterisk itself is not considered to be behind NAT.
17:28.41p3nguinBut any phones would be.
17:28.42Naikrovekif asterisk is compromised and gives a rootshell, attacker has access to your internal network
17:28.57Naikrovekp3nguin: yes
17:29.07*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
17:29.30*** join/#asterisk cusco (n=trilili@2001:0:53aa:64c:8b:771c:b2c9:8871)
17:29.32cuscohi
17:29.38[TK]D-Fenderpeer entries for Isps should almost always be nat=no
17:29.38p3nguinAnd since the phones are talking local addresses to/from Asterisk, I think local phones can also be set to nat=no.
17:29.42[TK]D-FenderITSPs
17:29.55Naikrovek[TK]D-Fender is correct; phones should be nat=yes
17:30.20QwellNaikrovek: They aren't behind a NAT
17:30.29Qwellrelative to Asterisk..
17:30.38Naikrovekcan the phones be reached directly from the internet?
17:30.41cuscoWe have several warnings like "Sep 16 18:16:33] WARNING[24867] file.c: File followme/ does not exist in any format"
17:30.57[TK]D-Fendercusco: Perhaps you should make the requested file exist...
17:31.39Naikrovekor point it in the right place
17:32.16user4545why Sipgate close DTMF
17:32.16cusco[TK]D-Fender: good point, tho we have manny followme's and can't find one that does not request the filename
17:32.20user4545?
17:32.22cuscoany ideias on how to debug that?
17:32.26p3nguinRelative to Asterisk the phone are not NATed.  Relative to the provider, the phones are behind NAT.  That was my entire reason for asking which way it should be configured.
17:32.52[TK]D-Fendercusco: Well we don't see what is being requested, or confirmation that the file it's looking for should be found, etc
17:33.05[TK]D-Fenderp3nguin: all as "no"
17:33.27Naikrovekall as no, really
17:33.40*** join/#asterisk el_critter (n=critter@200.8.96.143)
17:33.44el_critterHi
17:33.48NaikrovekiH
17:33.54p3nguinThat was my first thought, but I wanted an expert opinion before I left it that way.
17:36.00superbeefanybody dealt with this bug? http://pastebin.ca/1568596
17:36.18superbeefmakes the system load average go to the insanity point... like 120
17:36.35superbeefi saw a bug report for asterisk 1.6, but nothing too useful there
17:36.56superbeefthis is on 1.4.26
17:37.00Naikrovek120, that's respectable
17:37.14superbeeflol, i respect 1.0 more
17:37.31Naikroveki've seen some solaris boxes go up to 850
17:37.55NaikrovekNOT barf, then come back down
17:37.55superbeefwow
17:38.01Naikroveksame reaction i had
17:38.18Naikrovekbut no, i've not seen that.  does the bug report give any reason as to the cause
17:39.24Naikroveki was tech support at a library software company outside of chicago, customer called saying that their machine was unresponsive
17:40.28Naikroveki was able to ssh in, noted the 800 850 850 load average (after much waiting) then reload the appropriate service and it dropped down to 2
17:40.31superbeefi can't find the bug report now lol.. looking at some differnet threads
17:40.52Naikroveklady on phone said "ooh the server just got quiet"
17:41.04superbeefhaha
17:41.28superbeefhere's the one that seems most like mine
17:41.29superbeefhttps://issues.asterisk.org/view.php?id=15900&nbn=1
17:41.34Naikrovekyeah i kinda like tech support jobs for that reason.  over time, you see everything possible
17:41.57Naikroveksuperbeef: did your iax2 channel fail
17:42.21superbeeflogging goes dead after it gets saturated
17:42.23superbeefso hard to tell
17:42.42Naikrovekwhat fixes it?  asterisk restart or machine reboot
17:42.56superbeefkillall -9 asterisk
17:43.06Naikrovekhow often does it happen
17:43.11superbeefevery 2 hours
17:43.13Naikrovekeek
17:43.16superbeefyeah
17:43.20superbeefif not more frequnelty
17:43.25superbeefi just put this into production last night
17:43.31Naikrovekoh wow
17:43.41Naikrovekhard to take it out?  easy come easy go
17:44.01superbeefswap isnt too bad, but other box is asterisk 1.2
17:44.07dustybinis it possible to manually input contacts for a polycom phone using the web interface?
17:44.08Naikrovekwhat version is this one
17:44.22superbeef1.4.26-1 i think
17:44.31p3nguinI use 1.4.24.1 without any signs of trouble.
17:44.42Naikrovekdustybin: possible to manually input them via the phone, but not hte webui, or, you can put them into an XML file on the server and the phone will download them next time it reboots
17:44.46p3nguincore show version
17:44.59superbeefthose errors I have are ring groups calling quees in other PBXs via IAX
17:45.12dustybinNaikrovek: seems strange why they missed it off the web gui
17:45.33Naikrovekwell if this bug shows up for the bug reporter when an iax2 trunk fails, i would suspect something similar for superbeef
17:45.39dustybinNaikrovek: im not using tftp or ftp, so i will need to manually put them in... maybe i should start using ftp
17:45.53Naikrovekdustybin: i thought you were?  how many contacts are you going to add
17:46.02dustybinNaikrovek: not that many, about 15
17:46.08Naikrovekah
17:46.21dustybinNaikrovek: i configured the phone via the web interface
17:46.25Naikrovekdustybin: well the web interface is more for admin than daily use by the phone user
17:46.32dustybinaye ok
17:46.41dustybinmaybe its time to switch to ftp
17:46.49p3nguinAlso, is there any reason I will ever need canreinvite=yes?
17:47.54Naikrovekdustybin: i use ftp; it's nice.  the root of your ftp filesystem will need a polycom/contacts/ folder.  set up ftp, place that folder there, add a contact to the phone, reboot the phone, check the folder for a contacts.xml file and then you can see the format of an entry and add additional entries to that.
17:48.17dustybinthank you :)
17:48.21*** join/#asterisk gardo (n=gardo@121.97.136.60)
17:48.35[TK]D-Fenderdustybin: Didn't I tell you about this...
17:48.43dustybinhides
17:48.55[TK]D-Fenderdustybin: People configuring Poycom phones outside of a provisioning server should be dragged out an SHOT.
17:48.58superbeefI don't really know asterisk bug reportined ettitquite, can I respond to this bug report if i'm on 1.4.26?   https://issues.asterisk.org/view.php?id=15900&nbn=1
17:49.07[TK]D-Fenderdustybin: ... and survivors should be shot AGAIN
17:49.07dustybin:(
17:49.11jayteeyou can run but you can't hide from the [TK]D-Fender
17:49.26dustybin[FAILS]
17:49.29[TK]D-Fenderjaytee: That's why I go for the knee-caps first :p
17:49.45jayteehe's 1/4 bloodhound.....see!! his nose is cold!!!
17:49.47dustybin[TK]D-Fender: i didnt like the idea of installing a ftp server on my box just for 1 phone
17:50.04p3nguinThat's why I said use a tftpd.
17:50.16[TK]D-Fenderdustybin: yippy-kia-yay.  You probably already have one, or could in about 1 minute flat
17:50.29[TK]D-Fenderand no, TFTP = bleh
17:50.36jayteetftp sucks
17:50.53dustybini will install VSFTP
17:50.55p3nguinOdd.  Works for me and for most of Cisco.
17:50.58dustybini have used it ages ago
17:51.06jayteegood move, vsftp works awesome
17:51.15*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
17:51.19dustybinall the inforation is in the book
17:51.22dustybinreads
17:51.43jayteedustybin, have you read Polycom's whitepaper on configuration?
17:52.06dustybini guess running ftp on my local network isnt such a bad thing
17:52.12dustybinjaytee: no
17:52.28jayteehighly recommend you do
17:52.28dustybinok
17:54.28jayteedustybin, if you go here then halfway down the page the last item in the voice section titled "Configuration File Management on Soundpoint IP phones"
17:54.31jayteehttp://www.polycom.com/products/resources/white_papers/index.html
17:57.06dustybinthanks :D
17:57.36*** join/#asterisk Gnutoo (n=gnutoo@host98-153-dynamic.51-79-r.retail.telecomitalia.it)
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18:01.28*** join/#asterisk grandpapadot (n=no@99-175-248-81.lightspeed.brhmal.sbcglobal.net)
18:01.39*** join/#asterisk eduardovra (i=eduardov@189.42.32.228)
18:01.46grandpapadotHey guys, what's the state of presence in 1.6?  Does it support userstate via PUBLISH yet?
18:01.55*** join/#asterisk garymc (n=garymc@host86-173-16-209.range86-173.btcentralplus.com)
18:02.15[TK]D-Fendergrandpapadot: Not AFAIK
18:02.51grandpapadotTK: Thanks.  Are you guys using the users states with your Polycom's?  If so, how?
18:03.10Naikrovekgrandpapadot: yes, via openfire & spark
18:03.17Naikrovekjabber server & client
18:03.18[TK]D-Fendergrandpapadot: Not usable
18:03.30Naikrovekbut it's not a polycom specific solution
18:03.32[TK]D-Fender(with *)
18:03.35grandpapadotNaikrovek: Yea, that's easy, I'm talking about being able to set it via the polycoms.
18:03.42Naikrovekgrandpapadot: no idea
18:03.44grandpapadotTK: tnx
18:04.46*** join/#asterisk Meaty (n=meaty@office.abi.ca)
18:04.53*** join/#asterisk puzzled (n=foobar@puzzled.xs4all.nl)
18:05.06dustybinthe books example of setting up VSFTP is _terrible_ to say the least
18:05.57p3nguindustybin: It's simple.  I probably have some old confs if they would help.
18:06.22Naikrovekdustybin: use vsftpd.  two minutes you're up, and you probably already have it installed
18:06.26dustybini remember setting up VSFTP was quite a mission from past experience!
18:06.30Naikrovekno no no
18:06.31*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
18:06.31*** mode/#asterisk [+o putnopvut] by ChanServ
18:06.34Naikrovekwell
18:06.37Naikrovektook me 2 minutes
18:06.50dustybini did exactly what it said in the book
18:06.58dustybini created a user and group: PlcmSpIp
18:06.59p3nguinThat's about a minute longer than it should have taken.  You must have stopped to read the comments.
18:07.01raden_workis there a way to transfer a call and have some sort of status that the transfer went though and also when a call is transfered to have like a beep on the recieving parties line to let them know it transfered
18:07.09raden_workNaikrovek, howdy
18:07.23Naikroveksudo apt-get install vsftpd; vi /etc/vsftpd/vsftpd.conf; vi /etc/vsftpd/ftp_users.conf; /etc/init.d/vsftpd restart
18:07.32Naikrovekhowdy, raden_work
18:07.47Naikrovekp3nguin: i did, in fact
18:08.05p3nguindustybin: I sent you a link to my old confs in a /notice
18:08.51p3nguinThe default vsftpd.conf shouldn't need very much adjustment to get it going.
18:08.57Naikroveknope
18:09.03Naikrovekjust enable named users
18:09.18Naikrovekoh i forgot to "useradd phones; passwd phones"
18:09.30dustybinp3nguin: where did you send the link?
18:09.39p3nguinI normally have to add in that I want it to run as a daemon, too.
18:09.39Naikrovekdustybin: to your irc client
18:09.46dustybinim running irssi
18:09.51p3nguinLook on window 1.
18:09.53Naikrovekdustybin: check the server status window if you have one
18:09.54dustybinoh yeah
18:09.57dustybin:) thanks
18:10.25dustybinim running debian lenny, so maybe things go in different places
18:10.46*** join/#asterisk |Cybex| (n=John@80.100.126.176)
18:11.53*** join/#asterisk bluOxigen (n=asad@static-host119-73-71-157.link.net.pk)
18:12.24*** join/#asterisk tamiel (n=tamiel@ip-7.net-81-220-254.rev.numericable.fr)
18:12.25garymcCould logging into Zoiper and reging a SIP Softphone when im in the office on the same connection mess with the firewall on my router?
18:12.46garymcCos I got home now and I cant make calls to the office again?
18:13.32garymcI been getting calls from people in Texas and Isreal all day... again as soon as I get home its not working again :~(
18:14.49[TK]D-Fendergarymc: you weren't outside your work firewall all day, so you are comparing apples & oranges
18:14.54*** join/#asterisk seanmh (n=johndoe@207.114.199.107)
18:15.44garymc[TK]D-Fender : I wasnt no. But I had people in texas and Isreal and Tunisia login to my PBX as an extension and we made calls perfectly
18:16.16[TK]D-Fendergarymc: says something about your HOME setup, now doesn't it?
18:16.24garymcDoes it?
18:16.32Naikrovekgarymc: is your pbx on your laptop or something
18:16.44garymcNo its a HP proliant Server
18:16.52garymcin the office with a wEB server and an LTSP
18:17.03Naikrovekokay so tunisia and israel and texas can call each other still, just you that can't get in
18:17.15garymci dont know if they can now
18:17.30garymcbut they could today about an hour before i left the office
18:19.05garymcbut things did start acting weird, cos (i know this aint the channel) but i was logged into extension 202 with my laptop using zoiper. Then before I left my GUI was showing extension 202 logged in when i had unregistered Zoiper and shut down the laptop
18:20.02garymcit just kept showing as though the phone was there I could call extension 202 and it rang but there was no phone connected to 202?
18:20.18KavanSgarymc, are you trying to run soft phones on LTSP?
18:20.29garymcKavanS : NO
18:20.31garymcno
18:20.35KavanSok, my bad...
18:21.24garymcSo to me either my router firewall is deciding to not work after a few hours or possibly my GUI is messing with settings somehow?
18:21.56garymcbut i dont get why my GUI would show the third phone connected when it clearly wasnt connected
18:22.48*** join/#asterisk w9sh (n=chatzill@adsl-068-209-117-205.sip.asm.bellsouth.net)
18:22.59*** join/#asterisk propellerhead (n=yogurt2u@host26.190-137-6.telecom.net.ar)
18:23.01[TK]D-Fendergarymc: it doesn't miraculously change when you leave.
18:23.16garymcNo thats what i thought, but it is
18:23.22[TK]D-Fendergarymc: So either you specifically changed something after those successful calls, or your home setup is at fault
18:23.42[TK]D-Fendergarymc: and * can't FuBAR your firewall
18:23.55garymcno could Zoiper?
18:23.57*** part/#asterisk user4545 (n=sipip@dslb-092-074-252-179.pools.arcor-ip.net)
18:24.15garymcso why is freepbx showing 3 phones connected when only 2 where?
18:24.32[TK]D-Fendergarymc: go look at something real
18:24.42garymcwhat you mean?
18:24.47[TK]D-Fendergarymc: go look at something real
18:25.05dustybinat last, vsftp works with PlcmSpIp username and PlcmSpIp password :D thanks for help!
18:25.15dustybinthe book misses vital instructions
18:25.26[TK]D-Fenderdustybin: Don't.  Go setup a completely different suer & strong password
18:25.32[TK]D-Fenderdustybin: NEVER use those defaults
18:25.35dustybin[TK]D-Fender: its local only!
18:25.47[TK]D-Fenderdustybin: and just enter that in during the boot menu
18:25.47dustybin21 will _NEVER_ be open to the outside :D
18:25.55dustybin[TK]D-Fender: ok ..
18:25.56wonderworldwhich open-source iax-softphone would be the most advanced by now?
18:25.57garymc<[TK]D-Fender>garymc: go look at something real ------------< like what a car? what are you saying?
18:25.57superbeefare there any good load testing scripts for asterisk?
18:26.02[TK]D-Fenderdustybin: Doesnt' amtter.  If another PC gets hacked it opens your server to attack
18:26.14[TK]D-Fendergay&^$#ing CLI.  SIP DEBUG.
18:26.14dustybinaye good point
18:26.23[TK]D-Fendergarymc: &^$#ing CLI.  SIP DEBUG.
18:26.47garymcoh
18:26.50[TK]D-Fendersuperbeef: take a look at "sipp"
18:29.30wcselbyahhh
18:29.35wcselbypizza buffet
18:29.48wcselbyand I get a call right in the middle "hey we can't login to the conference bridge"
18:30.05*** join/#asterisk Skeeter- (i=Skeeter@190-141.cgocable.ca)
18:30.18wcselby"so and so had me change all the passwords, they have the spreadsheet with the new ones"
18:30.22*** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221)
18:30.22wcselby"well they're not heree"
18:30.22garymc[TK]D-Fender : This was home to office answer phone. I hear nothing I know its broken. http://pastebin.ca/1568668
18:30.42wcselbyme - "damnit!"
18:31.10superbeef[TK]D-Fender: thanks.....     How do you test T1 cards
18:32.30[TK]D-Fendergarymc: I see noone answered that call
18:32.49garymcyep the answer phone should have
18:33.05garymcI know its broken cos I dont hear the answer phone
18:33.06[TK]D-Fendergarymc: Nobody answered the phone...
18:33.21garymcbut shoouldnt i get the answerphone?
18:33.30[TK]D-Fendergarymc: WTF is "answerphone"?
18:33.37garymcVoicemail
18:33.40[TK]D-FenderGAH
18:33.50[TK]D-Fenderswats garymc with a disctionary
18:33.54garymcWhen it works I hear voicemail
18:33.56[TK]D-Fenderdictionary even
18:34.13[TK]D-Fendergarymc: Fine.  Maybe your home internet connection is crap
18:34.14garymcIm F&*KING ENGLISH!!! you know the language we all spek here
18:34.21garymc*SPEAK
18:34.22dustybin[TK]D-Fender: is there also a way to change the username from: PlcmSpIp  to something else?
18:34.37Naikrovekdustybin: yes
18:34.37[TK]D-Fendergarymc: "answerphone" is not a valid term
18:34.39dustybinace
18:34.49Naikrovek[TK]D-Fender: in britain it's acceptable, which is where he is
18:34.52dustybini will use username: polycom  password: strong
18:34.53Skeeter-~answerphone
18:34.57garymc[TK]D-Fender : My internet at home is better than office!
18:34.58dustybinand make sure the shell cannot login
18:35.01[TK]D-Fenderdustybin: rebooth the phone.  Enter setup.  Changer the boot server IP, user & pass threre
18:35.06dustybinthanks
18:35.16[TK]D-Fendergarymc: Apparently only FASTER.  But you might be getting FILTERED
18:35.18Naikrovekyup what [TK]D-Fender said
18:35.34[TK]D-Fendergarymc: Or your home router could be a flaming pile of shit
18:35.38garymcWell it worked last night when I got my Co worker to set the Router to DMZ
18:35.50Naikrovekthere's the clue
18:35.50garymcCalls worked fine
18:36.10garymcYeah DMZ. I tutrnt DMZ off today and set the ports and it worked all day!
18:36.49Naikrovekgarymc: that was at the office, yes?
18:37.02wcselbyno, I was able to login from MY office to HIS server (whereever that is)
18:37.09wcselbyand everythign worked just fine
18:37.13Naikrovekah
18:37.17wcselbygarymc, you want me to try to login again?
18:37.18Naikrovekbut not any more
18:37.20Naikrovek?
18:37.22*** join/#asterisk |omni| (n=rob@67.185.91.139)
18:37.31wcselbyNaikrovek - evidently.  i don't know, I went to lunch since then
18:37.33garymcNaikrovek : But I had people in texas, Tunisa and Isreal log in and we made calls to each other
18:37.37Naikrovekyeah
18:37.41*** join/#asterisk Takapa (i=vegard@junior.svanberg.no)
18:38.01|omni|anyone having a problem with dahdi channels under 1.6.1.5 and dahdi 2.2.0 ?
18:38.05*** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net)
18:38.05*** mode/#asterisk [+o mog] by ChanServ
18:38.05[TK]D-Fendergarymc: And you then left everything alone, went home, tried to log in the same as them and it doesn't work?
18:38.08wcselbyi wonder if I'm logging my chats
18:38.21wcselbyi can find the info to relog into his server
18:38.26|omni|I'm getting this crap [Sep 16 09:57:32] ERROR[10808]: chan_dahdi.c:10760 dahdi_pri_error: No more room in scheduler
18:38.26wcselbybleh, re-register with his server
18:38.26|omni|[Sep 16 09:57:32] ERROR[10808]: chan_dahdi.c:10760 dahdi_pri_error: Asked to delete sched id -1???
18:38.35|omni|on my PRI..and it completely stops responding
18:38.47garymc[TK]D-Fender : Just one thing..... Freepbx (I know other channel) was showing 3 phones connected when there was only 2
18:39.04Naikrovekgarymc: don't trust the FOP
18:39.04|omni|so..I came from zaptel..noticed that previous default was to reset PRI chans every hour..new dahdi default is not..so I have them reset
18:39.04[TK]D-Fendergarymc: Please confirm the exact situation I jsut told you
18:39.21|omni|works most of the time but usually in the a.m. after low or no call transaction is just dies
18:39.23garymcYes I touched nothing apart from make calls
18:39.50garymcbut the GUI started showing my Zoiper Extension connected when it wasnt
18:39.52[TK]D-Fendergarymc: then your home setup clearly ahs issues
18:40.01garymcno it doesnt
18:40.11garymcdefinatley my home setup doesnt
18:40.13[TK]D-Fendergarymc: Yes, it does
18:40.16garymcoh ok
18:40.25garymchow does it
18:40.39Naikrovek[TK]D-Fender: his independent office pbx stops working when he leaves; nothing else changes.  i think the timing is just a coincidence.  someone is unplugging yoru server I bet
18:40.45garymchave you got a test server I could make a call too from my home that you know works?
18:40.47wcselbygarymc - I just logged in using the info you gave me earlier, and dialed ext 201 - and went to voicemail
18:40.48[TK]D-Fendergarymc: You seem to have problems with the most basic aspects of the scientific process
18:41.11garymcwcselby : NO WAY!
18:41.13wcselbygarymc - well, it rang first, then went to voicemail
18:41.14Naikrovekwcselby: ah hah!  okay now we know that garymc's system is up
18:41.27wcselbygarymc - YES WAY
18:41.32wcselbygarymc- ;)
18:41.33Naikrovekgarymc: i'm with [TK]D-Fender now; your have outgoing ports blocked
18:41.44[TK]D-Fendergarymc: Everything works for other remote users.  You leave the server as it was.  You go home.  You places calls.  YOURS don't work.  Answer : your home setup is fucked up.  It is just you and you are in a pathetic state of denial.
18:41.45garymcshit! I hate it that [TK]D-fender has just .......... told me again!!!
18:42.00wcselbylol
18:42.05garymclol
18:42.12Naikrovekit does suck when you're mad and arguing and you realize that your point is incorrect
18:42.20Naikrovek[TK]D-Fender has shown me the way a few times
18:42.22garymcyes :#(
18:42.27Naikrovekeh, it happens
18:42.32[TK]D-Fendergarymc: And your home ISP could be completely fucking you over and it wouldn't even be a setting you have any control over
18:42.33Naikroveklearn from it
18:42.44garymcfukc sake
18:42.50Naikrovekmaybe he could SSH into his * server and do some tunnelling
18:43.00Naikrovekmaybe.. not sure if you can tunnel that many ports
18:43.05wcselby[TK]D-Fender knows what he's speaking of
18:43.11Naikrovekyeah he does
18:43.14wcselbyNaikrovek - you could do it with an iax phone
18:43.25*** join/#asterisk Katty (n=asterisk@mail.copi-rite.com)
18:43.28Naikrovekit sucks when he tells me i'm wrong when i'm sure i'm not, then it turns out i am wrong
18:43.31wcselbyiax is only one port to worry about, yes?
18:43.34garymcappologises for ever doubting the mighty [TK]D-Fender :P
18:43.38Naikrovekwcselby: yes
18:43.45[TK]D-Fendergarymc: ISPs that offer VoIP services as well often purposefully interfere with other VoIP services jsut to blackmail you into using theirs
18:43.53Naikrovekyeah
18:43.54Naikrovekthey do
18:43.59NaikrovekBT is bad about that I bet
18:44.03Naikrovekthey're a bunch of tools
18:44.07garymcpossibly
18:44.10Naikrovektried to patent hyperlinks a few years back
18:44.12Naikrovekidiots
18:44.27garymcI can only get BT in my area
18:44.36garymcat home
18:44.40[TK]D-Fendergarymc: go setup a VPN to the office and see if that works.
18:44.43garymcso we use BT in office too
18:44.47Naikrovekwell ask them if they're blocking things
18:44.48wcselbyyou could always call up and ask politely for them to open ports for you..........
18:44.52Naikrovekah yes vpn
18:44.59Naikrovekvpn vpn vpnvpn
18:45.01wcselby....
18:45.09wcselbylaughs quietly to himself
18:45.16[TK]D-Fendergarymc: And just because you use BT in both doesn't mean they get the same treatment either
18:45.42garymcyeah one is business BT and the other is Residential
18:45.48garymcfuk sake!
18:45.48Naikrovekyeah, comcast is like that.  business customers can do what they want, home customers have certain ports blocked so they can't do certain things
18:45.55[TK]D-Fendergarymc: and I never say you do a proper UDP port test across your range.
18:46.11[TK]D-Fendersaw*
18:46.12garymcmy home range?
18:46.12Naikroveki totally understand garymc's mood right now
18:46.18wcselbyat&t u-verse, you have to call and get 2nd tier support to remove the block on port 25 so that you can use an email server other than theirs
18:46.23[TK]D-Fendergarymc: both
18:46.32Naikrovekgarymc: grab .. ooh what's it called. .. nmap
18:46.36wcselbybut they will remove, if you convince them you aren't stupid and aren't going to become a spambot
18:46.48Naikrovekand see if you can scan your voip server from home
18:46.56garymcright ok
18:47.22Naikrovekscan TCP & UDP, ports 1024-20000.  it will take some time
18:47.38Naikrovekprobably overkill but you'll see what you need to see
18:48.17*** join/#asterisk sn00p- (i=sn00p@c-66-41-139-23.hsd1.mn.comcast.net)
18:48.27sn00p-DO I need any hardware for asterisk to work?
18:48.37superbeefIs DAHDI echo cancelation super resource expensive?
18:48.39Naikroveksn00p-: you need a linux box to run it on
18:48.55sn00p-Naikrovek, yea I know but do I need any phone hardware?
18:48.56Naikroveksuperbeef: probably done in hardware, no not super intensive
18:49.05sn00p-like a pbx switch or something
18:49.05Naikroveksn00p-: not necessarily
18:49.28wcselbysn00p- - you can do it all completely with a linux box running asterisk and a softphone
18:49.35superbeefNaikrovek: Sadly I have a Sangoma A101 and not the D, so software echo cancelation is enabled...   I'm still getting load averages much higher than I should for 5 or 6 calls
18:49.38*** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc)
18:49.42sn00p-Naikrovek, so is it possible for me to set up asterisk and be ale to send SMS?
18:49.46Naikroveksn00p-: you can use softphones and if you want to call out you'll need an IP voice provider
18:49.49wcselbysn00p- - but it's be a pretty small server.  it depends on what you want to do
18:49.51*** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc)
18:50.16sn00p-wcselby, all I want it to do is send SMS messages on the asterisk server
18:50.18Naikroveksn00p-: who are you trying to send SMS to
18:50.24sn00p-Friends
18:50.27Naikroveksn00p-: what provider(s)
18:50.35[TK]D-Fendersnoo* is NOT an SMS server
18:50.44[TK]D-Fendersn00p-: * is NOT an SMS server
18:50.45wcselbysn00p- - i haven't tried sending sms, but I think there's an app for that
18:51.03*** join/#asterisk thansen (n=thansen@76.27.110.194)
18:51.04[TK]D-Fenderwcselby: Only works on EU PRI's
18:51.25Naikroveksn00p-: there are pages online that you can use to do that.  usually providers have their own page where you can send to customers of that provider
18:51.25sn00p-wcselby, do you recall the app for SMS ?
18:51.38sn00p-Naikrovek, yea But I want my own
18:51.38[TK]D-Fendersn00p-: You have an EU PRI?
18:51.58Naikrovekhe's in minnesota, based on his hostname
18:52.22[TK]D-FenderNaikrovek: thats what it says alright...
18:52.33[TK]D-FenderNaikrovek: but that doesn't mean it counts...
18:52.36sn00p-So there is something seperate for sms server?
18:52.38Naikrovekso probably not... oh wait EU doesn't mean europe probably
18:52.46[TK]D-Fendersn00p-: Not Asterisk <-
18:53.02Naikrovekyeah this isn't something asterisk was meant to handle; not natively anyway
18:53.04wcselby[TK]D-Fender i thought I had read that chan_mobile will do that with a mobile phone connected via bluetooth
18:53.09[TK]D-FenderNaikrovek: And yes, I am asking if he had a European PRI
18:53.30Naikrovekany solution would involve some goofy hack that wraps around http posts or bluetooth phone
18:53.48wcselbyyeah, that
18:53.58*** part/#asterisk sn00p- (i=sn00p@c-66-41-139-23.hsd1.mn.comcast.net)
18:54.00*** join/#asterisk i9 (n=arthurh@70.56.139.60)
18:54.04*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
18:54.04*** mode/#asterisk [+o Deeewayne] by ChanServ
18:54.05[TK]D-FenderBAI BAI
18:54.08Naikroveknot even a goodbye or a thanks
18:54.13wcselbyhahaha
18:54.19[TK]D-FenderNaikrovek: Consider it a small mercy
18:54.21Naikrovekyeah
18:54.23Naikroveki do
18:54.25Naikroveki was happy abou tit
18:54.27Naikrovekabout it
18:54.35wcselbyyou're not happy about tit?
18:54.39Naikroveklol
18:54.45Naikrovekmy favorite bird
18:55.32Naikrovekhere's something that made me laugh, if anyone is bored: http://www.collegehumor.com/article:1791517
18:55.44wcselbyNaikrovek - you might be able to do something clever with google voice / jabber and asterisk
18:56.05*** join/#asterisk wathek (n=wathek@41.224.194.132)
18:56.09wathekhey all
18:56.14wcselbyjust an idea that popped into my head, no idea if it would work at all
18:56.32Naikrovekwcselby: for SMS?  could do IM that way perhaps
18:56.44Naikrovekwathek: yahey
18:56.47wathekis it possible to configure nicknames as extensions instead of numbers ?
18:57.03wcselbywathek - yes, but then what?
18:57.19Naikrovekwathek: i think so, but keeping names and numbers separated, uncoupled, is better
18:57.24wcselbywathek - you can do something like 'exten => nickname,1,NoOp()
18:57.29wathekwcselby, it's easier to keep in minde than numbers
18:57.40*** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw)
18:57.40wcselbywathek - you have phones that can dial names instead of numbers?
18:57.42Naikrovekwathek: how do you dial a name on a phone though
18:57.42wathekwcselby, ok thank you
18:57.52wathekNaikrovek, lol
18:57.56wcselbyi've heard some phones can do it
18:58.03wathekno but I'm just configuring just SIP
18:58.11wathekNaikrovek, yep I think it's possible
18:58.19ChainsawMy Cisco 7960s have a URL/Number softkey.
18:58.34wcselbywell, are you talking dialplan extensions or just sip usernames?
18:58.39wcselbybecause sip usernames is really simple
18:58.41[TK]D-Fenderwathek: if you aren'te expecting to use those in an IVR, sure...
18:58.58[TK]D-Fenderwcselby: You don't dial usernames....
18:59.19wathekwcselby, just SIP
18:59.28wcselby[TK]D-Fender - I know that, but he said he's just configuring SIP
18:59.29*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:e0b2:150:ec3e:2a25)
18:59.40Naikrovekcould do exten => 123,1,Dial(SIP/DaveyMakinCopies);
18:59.51wcselbywathek - ^^ what Naikrovek said
19:00.15wathekand then users could put DaveyMakinCopies to call that number ?
19:00.17Naikrovekthen you could name your devices, and use the exten => lines for extension to phone napping
19:00.23Naikrovekwathek: that, or 123
19:00.28wathekcool
19:00.29wathekthank you
19:00.37wcselbyin your sip.conf file, instead of naming the phones [201], you can name them [wathek].  then you dial using Dial(SIP/wathek)
19:00.49Naikrovekyeah
19:00.53wathekok
19:00.55wathekthat's cool
19:01.06wathekhave to find now how to limit the call duration
19:01.33wcselbyas in, once a call reaches it's limit you want to cut it off?
19:01.39Naikrovekthat would actually make sense in a way, you could keep track of MAC addresses that way.  [0004abcdef10]  - then exten => 444,1,Dial(SIP/0004abcdef10);
19:01.55wathekwcselby, yep
19:02.23Naikrovekhow long do you want to limit to
19:02.34wathek10minutesz
19:02.40wcselby[TK]D-Fender - didn't you help someone with that the other day?
19:03.16i9Hey guys, we're looking for a small group of closed beta testers for the latest internal beta of iSymphony -- with Asterisk 1.6 support -- if anyone's interested in participating, shoot an e-mail to isymphony-beta@i9technologies.com with environmental specs..
19:03.34Naikroveki9: would love to but i broke it on my laptop
19:03.38Naikrovekclient won't launch anymore
19:03.40Naikrovekyay
19:03.49Naikrovekwon't uninstall completely either
19:03.50Naikrovekyay
19:03.53Naikrovekheh
19:03.55i9Naikrovek, want some help?
19:03.56wcselbyNaikrovek - lol
19:04.00Naikrovekno, i gave up on it
19:04.03wcselbyi remember that day
19:04.08Naikroveki'll install it in a virtual machine and play there
19:04.27Naikrovekif i ever get any time
19:04.27wcselbywathek - limit call durations - http://forums.whirlpool.net.au/forum-replies-archive.cfm/776539.html
19:04.31wathekwcselby, call thank you so much
19:04.57*** join/#asterisk flohack (n=fhackenb@84.115.131.198)
19:05.05[TK]D-Fenderwcselby: Yup, that would be yesterday morning
19:05.13wcselbywathek - those are for older versions of asterisk, but should get you started on the right path
19:05.19i9Naikrovek, it's all self contained.  86ing the directory and a registry entry should get rid of it in it's entirety -- give us a ring or shoot an e-mail and we'll be happy to help.. we just need 1.6 testers at the moment
19:05.24wathekwcselby, ok
19:05.53Naikroveki9 yeah it's based on eclipse, which i'm very familiar with.  will have to nuke the registry and try again then
19:06.18superbeefSo does asterisk spawn a process for every phone call?
19:06.29i9Naikrovek, No, the ONLY registry entry we put in there is for the uninstaller (to appear in add/remove programs)
19:06.31wcselbysuperbeef - i don't think so.....
19:06.32Naikroveksuperbeef: don't think so...
19:06.37superbeefHmm
19:06.39Naikroveki9: okay
19:06.41i9Naikrovek, so not nuking the registry, just removing that entry
19:06.47Naikroveki9: yes i know that
19:06.48superbeefwell i have liek 15 asterisk instances running
19:06.56Naikroveki9: you keep plugging for beta testers i'll work on this
19:06.58wcselbyi9, Naikrovek, lol.  I think you both had this convo like two weeks ago
19:07.08superbeefmore like 20
19:07.09Naikrovekthat was seanmh, not i9
19:07.18i9haha..
19:07.30wcselbyNaikrovek - ahh, lol.  same convo as last time though
19:07.35Naikrovekpretty much
19:07.37[TK]D-Fenderi9: I suspect if you help Naikrovek with his issues he might be happy to give it a look...
19:07.38seanmhI'm here :D
19:07.43i9Naikrovek, No more plugging -- one post is enough I think in #freepbx, #asterisk and #trixbox ;)
19:08.10Naikrovek[TK]D-Fender: i'm CERTAIN that my issues are my own fault
19:08.18wcselbyi9 - using 1.4.26.  Will probably be building a 1.6 cluster in a couple months.  how long do you plan on running your beta?
19:08.23Naikrovek[TK]D-Fender: i just need to figure out what is going on and fix it
19:08.30[TK]D-FenderNaikrovek: And I'm still pretty sure you'd be happy for them to be solved :)
19:08.34Naikrovekyeah
19:08.39Naikrovekbut low on priority list
19:08.41[TK]D-FenderNaikrovek: My way both of you win
19:09.02i9wcselby, It's likely we'll be out of beta in a few months, General release should be within 30 days.. even if you're running 1.4, hop in.. there's quite a bit of bug-fixing and new stuff..
19:09.22wcselbyi9 - i'll send an email when I get a chance this week.  :)
19:10.15i9wcselby, awesome -- note in the e-mail that you're an irc contact and we'll throw you up top
19:10.44*** join/#asterisk mnicholson_ (n=mnichols@nat/digium/x-bnbwwkrzpnjhwcxt)
19:10.44*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
19:12.01*** join/#asterisk The_Boy_Wonder (n=davidvos@asterisk/batman-developer/dvossel)
19:12.02*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-okopfrwoucuizora)
19:12.40[TK]D-FenderOMG DIGIUM RAIDING PARTY!
19:13.08[TK]D-Fenderhides his contraband elecom gear
19:13.11[TK]D-Fendertelecom*
19:14.04Kattyoh man, the sleepies are getting me
19:14.56[TK]D-FenderThe Dream Police... reside in my head!
19:15.17*** join/#asterisk andres833 (n=andres83@190.144.75.22)
19:15.29*** join/#asterisk tzafrir_laptop (n=tzafrir@212.179.75.202)
19:16.16Naikrovekisymphony people: how the hell does admin/secret not work when: <SystemAdmin userName="admin" password="secret"/>
19:17.25Naikrovekno time to investigate this right now; i suppose i'll call one day
19:17.40Naikrovekthought i'd try again and maybe the gods i don't believe in would smile on me today
19:17.43Naikroveknope
19:18.31dustybini keep on getting: Could not contact boot server  on my polycom
19:18.37dustybinthe menus are confusing
19:18.50dustybinhowever, i did select ftp and put in my ip and username + password
19:18.59Naikrovekdustybin: your authorization is wrong, or your ftp server isn't running
19:19.09[TK]D-Fenderdustybin: And confirmed that your server matches?
19:19.16[TK]D-Fenderdustybin: logged in with another client?
19:19.37Naikrovekyeah use cmdline ftp client to test
19:19.45Naikrovekuse correct hostname, username, and password, verify that it works
19:22.03seanmhNaikrovek: hover over the server icon in the administration section next to server and let me know what it says
19:22.13garymcRight heres some news. BT reckon they arnt blocking nothing and i need to unblock the ports on my home HUB!
19:22.17*** join/#asterisk raden (n=jon@69.179.99.17)
19:22.29garymcso doen that and still not working :S
19:22.31*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
19:22.44[TK]D-Fendergarymc: And its still just you.
19:22.57[TK]D-Fenderspins up some Eric Carmen, jsut for garymc
19:23.25jayteeKatty, it was probably from the sour cream in the stroganoff
19:24.25*** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net)
19:24.27Kattyjaytee: it was the cream cheese, actually (=
19:24.40wcselbygarymc - have you tried using xlite to connect from home, just to be sure?
19:24.48wcselbygarymc - that's what I was using to connect to you
19:25.39jayteeKatty, I follow a similar recipe to the one you posted but I use a can of mushroom gravy and a can of Campbell's Beefy Mushroom soup instead.
19:26.19[TK]D-Fenderwcselby: Does it still work for you trying to call in to his office?
19:26.38wcselbyit did oh....i dunno, 20 minutes ago?
19:26.38p3nguinHe's still on that?  Wow.
19:26.52[TK]D-Fenderwcselby: Ok, that confirms it
19:27.33wcselbyand it still works now
19:28.00*** join/#asterisk merkurie (n=merkurie@192.153.163.45)
19:28.05Kattyjaytee: yum. i might try that for some additional deep flavor (=
19:29.14merkurieanyone else ever run into a problem where they have a router, with some sip devices inside behind nat, asterisk server on the public internet side, but the router always hands out the same source port number when the sip clients connect to ast/udp/5060? even though the sip clients all have different ips inside nat?
19:29.32*** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net)
19:29.41jayteeKatty, when I make it I often go all out and substitute sirloin tips I've broiled rare on the grill and then cut into cubes instead of ground beef/turkey
19:29.55ayesoAnyone using SS7 with asterisk?
19:29.57Chainsawmerkurie: Some routers try to be clever about SIP and rewrite packet headers.
19:30.09Chainsawmerkurie: You may have to disable this "feature".
19:30.41*** join/#asterisk Micc (n=dotirc@c-98-225-59-171.hsd1.wa.comcast.net)
19:31.25*** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net)
19:31.37merkurieChainsaw, dd-wrt?
19:31.40Kattyjaytee: that sounds delicious
19:31.46dustybingrrrrr because i put bin/false on the end of the passwd file for polycom, it failed
19:32.04jayteewe really need to build a comprehensive list of router manufacturers whose products try to rewrite SIP packet headers so that we can more efficiently purge them when the violent phase of the revolution begins
19:32.12Micccan I make two mailboxes notify a single sip account?
19:32.24Chainsawmerkurie: I don't know every router that was ever made by heart. You'll have to check your settings and see whether anything involves SIP header rewriting.
19:32.29p3nguindustybin: You sure that's why?  You shouldn't need a login shell to get files from the ftpd.
19:32.30dustybinhow can one reboot a polycom 321 without keep on pulling out the power cord?
19:32.47dustybinp3nguin: it needs /nologin,  not /false
19:32.57merkurieChainsaw, kk, thanks
19:33.04wcselbydustybin - isn't there  a restart command in the settings menu?
19:33.12wcselbymaybe advanced settings?
19:33.17wcselbyi know the 601 has one
19:33.23jayteedustybin , you can reboot from the Menu key
19:33.59p3nguindustybin: I doubt that's the case.  Chances are that your false shell just wasn't listed in /etc/shells.
19:34.06p3nguindustybin: Try pressing the 4, 6, 8 and * keys simultaneously.
19:34.26jayteedustybin, press Menu, choose Settings then Advanced (enter password) and then option 3 Restart Phone
19:34.57Naikrovekdustybin: menu, 3, 1, 4, yes softkey
19:35.16dustybinthanks :D
19:35.36dustybinhere we go, hopefully it will use ftp now
19:35.57dustybini've named my polycom 'sod'
19:35.58dustybin:D
19:36.03*** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker)
19:36.13dustybin'could not contact boot server..'
19:36.19dustybinchecks ftp with a client again
19:36.20wcselbyMicc - if you find out, let me know...but i think it depends on the type of sip device you want to notify
19:36.25dustybinshit i changed the password
19:36.27dustybinof course
19:36.31Naikroveklol
19:37.28dustybinits _SO_ easy to make a mistake, one tiny little error
19:37.35Naikrovekyeah
19:37.36dustybinone tiny digit
19:37.41Naikrovekbut once it's set up it'll work forever
19:37.46Naikrovekuntil you change a passwd again
19:37.53p3nguinlike four bits or so?
19:37.58dustybini think system admins need to pay close attention to detail if they want to be good admins
19:38.05bmoracadustybin: the phone will also give you that error if it doesn't find a file it can download, such as its MAC.cfg file
19:38.26Naikrovekbmoraca: true, but he can see what it is looking for via the ftp server log
19:38.35bmoracaright
19:39.00wcselbybmoraca - you read earlier that I was able to get 2 softphones over my at&t 2wire?
19:39.06wcselbyfrom 2 separate computers
19:39.29*** join/#asterisk juanIMP (n=juan@200.71.41.254)
19:39.37bmoracawcselby: i can get them to register fine, but they do not stay registered.  after 30 seconds or so, they are no longer available for incoming calls
19:39.51bmoracaor, rather, 5 minutes might be a closer estimate
19:40.01wcselbyhmmm....okay, I didn't last that long
19:40.08wcselbymaybe a minute or so total?
19:40.08bmoracathey work for 5 minutes, but then the 2wire loses its NAT and the phone no longer worked
19:40.23wcselbyi got both hooked up, then dialed out on both, which both worked.
19:40.29wcselbythen I shut down, cause I was going to bed
19:40.39wcselbyi'll try again tonight, let them run
19:40.42p3nguinSounds like you need a new network appliance if NAT stops working every five minutes.
19:40.44wcselbyovernight, then test in the morning
19:40.55bmoracadialing out was never the issue...it was dialing IN to the phone that was the issue
19:41.14bmoracap3nguin: indeed.  this is why i was complaining about SOHO gear last night (one of the reasons, anyway)
19:41.17wcselbyi know - i'll check that also
19:41.28*** join/#asterisk doolittlework (n=f@196.211.34.2)
19:41.53p3nguinIf NAT stops working, that means computers are losing their ability to connect to the internet, as well.  Correct?
19:42.12ayesop3nguin: no
19:42.23bmoracap3nguin: read what i wrote again.  NAT doesn't stop working, the router just does not preserve the phone's NAT
19:42.39garymcwcselby i will try xlite now
19:42.42p3nguinThat wasn't what I saw in the above-listed text.
19:42.49*** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7)
19:43.27p3nguin"then the 2wire loses its NAT", where 2Wire indicates the network appliance.
19:43.33bmoracawcselby: FWIW, the 2wire gateway for Uverse doesn't seem to have this issue.
19:43.45bmoracap3nguin: right, however "its" refers to the phone
19:44.00p3nguinAh, improper grammar usage caught me again.
19:44.48bmoracathere was nothing improper about my grammar.  within the context of my statement, it was very clear that i was talking about phones
19:44.48p3nguinI beg to differ.
19:44.48wcselbybmoraca - then I won't be much help, I've got at&t uverse
19:45.27p3nguinRegardless, I won't be of any help with the situation, so I'll refrain from additional comment.
19:45.33bmoracap3nguin: whatever.  you have fun with that.  meanwhile, i'll have fun making money off my hosted pbx customers.
19:45.42p3nguinlaughs
19:45.49*** part/#asterisk deeperror (n=deeperro@adsl-76-226-149-104.dsl.sfldmi.sbcglobal.net)
19:46.24bmoracawcselby: i've got uverse at home as well, and that 2wire seems to be far superior to the original 2wires AT&T sold, which is what this particular customer has
19:46.33*** join/#asterisk war9407 (i=war@liquidswords.org)
19:46.57wcselbybmoraca - :(
19:47.01bmoracawcselby: however, i sold him on a new router and a discrete DSL modem, which should resolve the issue.
19:47.11wcselbycool :)
19:47.58wcselbyone of the AA's at my client wants to buy a CS70N headset - anyone ever use this?  (from plantronics)
19:48.20wcselbywell, anyone ever use this successfully with asterisk / polycom phones ?
19:48.34bmoracanot that one in particular, but plantronics tend to be very good
19:48.49wcselbyyeah, they usually buy the CS55s here
19:49.01bmoracawcselby: if you have doubts about compatibility, tell them to get the one with the lifter and it'll be a non-issue
19:49.11wcselbyyeah, that's what they're doing
19:49.13wcselbycool
19:49.35bmoracafrom what i remember, though, polycoms tend to be compatible with almost any headset...Ciscos, on the other hand, are not
19:49.54p3nguinwcselby: It has a good customer rating, if that means anything.
19:50.36jayteewe use the Jabra GN2010 ST headset but it's not wireless, it uses a cord
19:51.03jayteebut it works great on Polycom 330's and 550's
19:51.05p3nguinHmm.  Only one person rated it on plantronics web site, though.
19:51.39*** join/#asterisk evil_gordita (n=evilgord@ip24-254-160-77.rn.hr.cox.net)
19:52.06MeatyHi Every Body!
19:52.09MeatyI want use Realtime SIP with ODBC.  I have set configurations in extconfig.conf and res_odbc.conf.  I add some rows in table with type = peer.  When i make "sip show peers" i have only my peers in sip.conf.  But when i make "realtime load sippeers username test", i see the colums values for my peers with username "test". Am I suposed to see the peer "test" when i make  "sip show peers" ?
19:54.20*** join/#asterisk [netman] (n=netman@216.Red-88-17-240.dynamicIP.rima-tde.net)
19:54.51wcselby<Meaty> Hi Every Body! <---- Hi Doctor Nick!
19:55.05wcselbysorry, i can't help with your issue
19:56.14*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
19:56.16doolittleworkhi there i am stuck with the monitor command, [TK]D-Fender said i must use "g" in the dial command as an option to continue with the dialplan if the called party drops the channel. cant seem to get this to work i just get a ingage tone. on the extension that made the call, any help welcom
19:56.20doolittleworke
19:57.53wcselbydoolittlework - please pb the relevant parts of your extensions.conf and also the cli output of a call that exhibits what you're talking about
19:57.59wcselby~pb
19:57.59infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
19:57.59dustybinin the polycom SETUP menu, i am following DHCP MENU > Boot Server: Static
19:57.59[TK]D-Fenderdoolittlework: show me the FAILURE
19:58.20dustybinthere is also a option for: IP Gateway ?
19:58.43[TK]D-Fenderdustybin: Not DHCP....
19:59.01dustybinthis menu is confusing to navigate
19:59.11*** join/#asterisk ebroad (n=EB@72.11.213.195)
20:00.10dustybin[TK]D-Fender: i need DHCP enabled so the phone is given a IP address?
20:00.28*** join/#asterisk el_critter (n=critter@190.78.48.45)
20:00.34p3nguinmakes sense to me
20:00.34dustybinthe first option is: DHCP Client: Enabled
20:00.41dustybinif i press DOWN
20:00.48dustybinDHCP Menu
20:00.53SuPrSluGany way to tone down a moh file in musiconhold.conf ? the files are wav
20:00.56dustybindown again
20:00.59dustybinIP Gateway
20:01.04dustybin00.00.00.00
20:01.11dustybindown again
20:01.14dustybinServer Menu
20:01.16SuPrSluGsimilar to quietmp3
20:01.31p3nguinIf you're getting an IP address via DHCP, wouldn't that configuration contain the gateway setting?
20:01.52dustybinp3nguin: yes it should
20:02.10dustybini am entering the Server Menu
20:02.17dustybinServer type: FTP
20:02.31dustybinAdress: = my server IP
20:03.00wcselbyhttp://austin.craigslist.org/cpg/1377692273.html <--- lol
20:03.07[TK]D-FenderSuPrSluG: Either use mpg123, or resample them yourself
20:03.51SuPrSluGk, thanks.
20:04.11maouri want to present somewhere about ip/pbx , anyone has any document/link/pdf/slide !!?
20:05.02[TK]D-Fendermaour: www.google.com
20:05.10[TK]D-Fender~wikis
20:05.11infobot[~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.  http://www.voip-info.org (c) Arte Marketing Inc / CommPartners
20:05.11ayesoheheh
20:05.13[TK]D-Fender^
20:05.16maour:)
20:05.30maouri mean some thing like a presentation
20:05.50maouri googled a lot ! nothing usefull
20:06.32Corydon76-digWhy don't you write your own?
20:06.43[TK]D-Fendermaour: http://www.google.ca/#hl=en&q=asterisk+pbx+presentation+powerpoint&meta=&fp=b664983305b547ce
20:06.43Corydon76-digThat's what the rest of us have to do
20:07.04maourHm! :D
20:07.15dustybinwould the polycom give the same message: could not contact boot server, even if it meant there was something wrong with the .cfg files?
20:07.23doolittleworkwcselby:http://pastebin.com/m4c789f44
20:07.51*** join/#asterisk ajohnson (n=ajohnson@65-122-4-130.dia.static.qwest.net)
20:08.11Corydon76-digI don't relish the idea of somebody else taking my presentation that I worked hard to create, changing only the author name, and presenting it as their own
20:08.51[TK]D-Fenderdustybin: has nothig to do with the cfg's
20:08.52wcselbydoolittlework - so once the called party hangs up, you go to 4,Hangup.....?  show me a log from the cli of a call doing what you don't think it should be doing....
20:09.23doolittleworkthe first one works records the file but http://pastebin.com/m7cf6151 this i swhat i did to see if the calls go to next step in dialplan
20:09.39dustybin[TK]D-Fender: what im trying to say is, if the phone did log into my ftp server, however, it couldnt find a config, would it give a error message saying: could not locate config ?
20:09.44NaikrovekSuPrSluG: you can use sox to normalize to a lower volume i believe
20:09.56doolittleworki should hear the vm-busy message but i just get ingage tone
20:09.58wcselbydustybin - check your /var/log/vsftpd.log
20:10.03NaikrovekSuPrSluG: if it supports mp3 on your system.  otherwise use lame probably
20:10.04[TK]D-Fenderdustybin: check your perms,e tx
20:10.11ayesoUsing asterisk with SIP for signaling. Can I use a separate interface for media then the one used for signaling?
20:10.12wcselbydoolittlework - i still need to see the cli output from during a call
20:10.26dustybinvsftp is not logging for some reason
20:10.27SuPrSluGNaikrovek: mp3 = too much overhead.
20:10.36wcselbydustybin - is it running?
20:10.38Naikrovekoh sorry misread
20:10.42dustybinohh it is!!!!!!!!!
20:10.42Naikrovekyes sox will do it for you
20:11.15wcselbydustybin - from a computer on the same network as your phone, try to login to the ftp server using the same credentials you're putting in for the phone.
20:11.15[TK]D-Fenderayeso: No
20:11.29doolittleworkk i just setting up putty sesion to box
20:11.52SuPrSluGactually I called and it sounds fine. they must be whiners
20:12.01ayeso[TK]D-Fender: didn't think so...
20:12.31wcselbySuPrSluG - I had to resample some wav files for a client once, they said it was too loud.  sounded fine to me, they wanted it lower so it went lower.
20:12.38dustybinmy polycom has made communication with VSFTP :D
20:12.50wcselbySuPrSluG - then they said it was too low, so I had to go back up.
20:12.50dustybinmy god, its left some new files behind
20:12.56wcselbydustybin - lol
20:12.58[TK]D-Fenderdustybin: So far so good
20:13.10wcselbydustybin - that's what polycom's do
20:13.27Naikrovekdustybin: that's how you check the phone log
20:13.31Naikrovekit uploads the log to the ftp server
20:13.40Naikrovekas well as any contacts (which i think is what started all this)
20:13.47SuPrSluGthey can send their own files if they want it. this is multi-tennant
20:14.27dustybini can see the problem now
20:14.36dustybinits full of FAIL
20:14.43Naikrovekwhoa there, cowboy
20:14.48Naikrovekyou smacktalkin' my polycoms
20:14.49wcselbyhaha
20:14.50Naikrovek?
20:15.35SuPrSluGdustybin: <mac>.cfg tells polycom phones where to find sip.cfg and phone.cfg files and directory.xml if desired.
20:15.53dustybinmy configs are named wrong
20:15.58dustybini should be using -directory.xml
20:16.08dustybin-license.cfg
20:16.29dustybinbootrom.ld' FAILED on attempt 1 (addr 1 of 1)
20:16.35dustybinim lacking a good howto
20:16.40dustybinagain, google was wrong
20:16.43Naikrovekdustybin: we are your howto
20:16.45SuPrSluGif defined in <mac>.cfg it doesn't matter what you call em
20:16.54Naikrovekwell the filenames for the firmware matter
20:17.18dustybindo i need to download something from here:
20:17.19dustybinhttp://www.polycom.com/support/voice/soundpoint_ip/soundpoint_ip330_320.html
20:17.24dustybineven though i have a 321
20:17.37[TK]D-Fenderdustybin: What did you extract into your provisioning folder?
20:17.39Naikrovekdustybin: download the split VVX SIP firmware and bootrom for your model phone, just dump those files into the root directory of your ftp server (the directory where everything else is)
20:18.09dustybin[TK]D-Fender: not a lot at the moment, just some rough configs, i will delete them now
20:18.38SuPrSluGdustybin: a mac.cfg should look like http://pastebin.com/m687e0a95
20:18.40[TK]D-Fenderdustybin: Sounds like a good reason not to find firmware there...
20:18.52dustybineeek
20:18.56[TK]D-Fenderdustybin: http://downloads.polycom.com/voice/voip/sip_sw_releases_matrix.html
20:19.36SuPrSluG321's have the lastest firmware no need to upgrade firmware
20:19.37[TK]D-Fenderdustybin: Go download and extract http://downloads.polycom.com/voice/voip/sp_ss_sip/spip_ssip_vvx_3_1_3RevC_release_sig_combined.zip
20:19.51[TK]D-Fenderdustybin: unless your phones hav 3.2.0
20:19.51dustybini will wget it into the ftp root
20:20.06SuPrSluG3.2.0 is out?
20:20.21[TK]D-FenderSuPrSluG: last month
20:21.01dustybin[TK]D-Fender: im not sure if my phone is 3.1 or 3.2
20:21.31[TK]D-Fenderdustybin: menu>status>platform>application>main
20:21.33SuPrSluGmenu -> status -> platform
20:21.59dustybin3.1.3.0507
20:22.28dustybinthat is a big .zip file
20:22.37[TK]D-Fenderdustybin: dl the fil I gave you and extract to your provisioning folder.
20:22.42[TK]D-Fenderdustybin: and we start from there
20:22.54dustybinok :D
20:23.50doolittleworkwcselby: just bare with me my networkcard on server is a bit dodgy
20:24.17wcselbydoolittlework - k, but I'm only here another 20-30 minutes
20:24.20dustybinok its unzip
20:24.56dustybinimages of jellyfish
20:24.57Kobazhow do i 'lock' CALLERID(name)... i accept a call, and then do a bunch of processing... by the time i actually send the call to a phone via Dial... the callerid name get's clobbered by something else
20:24.58dustybinstrange
20:25.07Kobazthis is for inbound calls on dahdi via t1
20:25.26wcselbyKobaz - are you messing with CALLERID(name) at all during processing?
20:26.21Kobazwcselby: yes, i set it
20:26.37Kobazi force callerid name depending on some factors
20:26.41Kobazbut i set the name
20:26.50Kobazand then dial the phone... and then the phone gets Unknown
20:26.58wcselbyhmmm
20:27.01Kobazit only happens when i get a call with a blank callerid name
20:27.07wcselbypb relevant parts of your extensions.conf
20:27.10*** join/#asterisk s519 (n=steve@87-194-151-213.bethere.co.uk)
20:27.14dustybin-r--r--r--  1 polycom polycom 68467899 2009-05-27 23:00 sip.ld   <-- is this the firmware
20:27.21wcselbywell if the callerid(name) is blank, then what's wrong with Unknown?
20:27.57SuPrSluGdustybin: yes
20:28.29wcselbypb relevant parts of your extensions.conf
20:28.38Naikrovekdustybin: if you downloaded the combined firmware
20:28.40SuPrSluGthere are also sip.ld for each phone model
20:28.43Naikrovekdustybin: split works better
20:28.48*** part/#asterisk korihor (n=korihor@190.77.83.180)
20:28.51Kobazhttp://pastebin.ca/1568840
20:29.08Kobazwcselby: the name has to be the name i set it to be... not unknown
20:29.10Kobazi have a name
20:29.19Kobazi set the name in CALLERID(name)... but it doesn't 'take'
20:29.28Kobazand it happens only when i get an empty callerid name
20:29.39p3nguinHow would I go about telling Cisco 7900 series phones to pull files from a subdirectory of the tftpd rather than the root of it?
20:30.06Naikrovekp3nguin: not sure you can
20:30.45[TK]D-FenderKobaz: no quotes
20:30.55Kobaz[TK]D-Fender: k
20:31.21wcselbyKobaz - let me see your extensions.conf where you're setting it
20:31.58wcselbybut [TK]D-Fender is probably right
20:32.21bpgoldsb'MYSQL(Fetch fetchid ${resultid} var1 var2 ... varN) -- Fetches a single row from a result set contained in ${result_identifier}. Assigns returned fields to ${var1} ... ${varn}.  ${fetchid} is set TRUEif additional rows exist in result set.'
20:32.32Kobazhttp://pastebin.ca/1568845
20:32.36wcselbyp3nguin - i don't think you can
20:32.47bpgoldsbI'm getting fetchid set to true, even though I'm only getting 1 row back.  shouldn't that not be the case?
20:33.16jayteeTroy Dale West of Poulan, GA is a mullet-wearing brain dead inbred redneck racist piece of shit. Just had to get that off my chest.
20:33.33Kobaz[TK]D-Fender: i've never had a problem using quotes
20:33.40Kobaz[TK]D-Fender: i don't think the quotes are the problem
20:33.43p3nguinI seem to recall seeing someone's post on some site about using "local/<configs>" rather than the root, but I don't know what to search for to find the correct info about doing it.
20:33.55[TK]D-FenderKobaz: you no longer DIAL or do anything else productive in that
20:34.05[TK]D-FenderKobaz: and I don't see the FAILED ATTEMPT with back
20:34.06Kobaz[TK]D-Fender: that's later
20:34.07[TK]D-Fenderup
20:34.08*** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek)
20:34.13Kobaz[TK]D-Fender: it's not a failed attempt
20:34.23[TK]D-FenderKobaz: I know.  What aren't I seeing it?
20:34.26Kobaz[TK]D-Fender: like... the dial is successful
20:34.27[TK]D-FenderWhy*
20:34.30Kobazheh
20:34.39Kobazfind fine... it's not useful
20:35.02jayteequittin time
20:37.32wcselbywho the hell is Troy Dale West?
20:37.54*** join/#asterisk grEvenX (n=even@cB78D5AC1.dhcp.bluecom.no)
20:38.25Naikrovekwho the hell cares?
20:38.36wcselbyevidently jaytee does
20:38.51SuPrSluGkanye's forgotten twin
20:39.13Naikrovektwin?
20:39.20Naikrovekdual douchebaggery?
20:39.20wcselbyhttp://www.ajc.com/news/clayton/army-reservist-beaten-in-138917.html?imw=Y
20:39.22wcselbyahh
20:40.02Naikrovekoh what an ass, a man, beating a woman, yelling slurs, in front of her daughter
20:40.10wcselbyyeah, i agree
20:40.15Naikrovekthere are fewer things you can do that are more traumatic to a child
20:40.19Naikrovekvery few
20:41.20Naikrovekin my utopia people like that are shot on site
20:41.31Kobaz[TK]D-Fender, wcselby : http://pastebin.ca/1568858
20:42.58[TK]D-FenderKobaz: where do i see you NoOp-ing it after having set it?
20:43.08Kobazhmm
20:43.10Kobazintersting
20:43.12Kobazi'll try that
20:44.13*** join/#asterisk AndyML (n=AndyML@pool-173-49-144-213.phlapa.fios.verizon.net)
20:44.22[TK]D-Fenderreaches for his ClueBat (tm)
20:44.42AndyMLso - can someone have a conversation with me about the differences in the different 1.6.X releases?
20:45.02[TK]D-Fender~asteriskversioning
20:45.02infobotasteriskversioning is, like, Information about the new Asterisk versioning method with the 1.6.x series is available here:  http://www.asterisk.org/node/48602
20:45.02dustybinIT WORKED!!!!!!!!!!!!!!!!!!
20:45.06[TK]D-Fender^^^^^^^^^^
20:45.17AndyMLit looks like there is a 1.6.0.current and 1.6.1.current. - thanks [TK]D-Fender
20:45.29AndyMLare you using either [TK]D-Fender?
20:45.32Kobaz[TK]D-Fender: heh
20:45.43[TK]D-FenderandI'm on 1.6.0 at home
20:45.49[TK]D-FenderAndyML: I'm on 1.6.0 at home
20:45.53Naikrovekdustybin: gratz
20:46.02AndyMLk. i'll read this and come back. tnx
20:46.06Kobaz[TK]D-Fender: i was having issues before... i had to insert a Wait() when processing the call...because i would answer the call... and then pri q931 callerid name would come in, and clobber the name
20:46.13dustybinTHANKS!! Naikrovek your configs work perfectly
20:46.20Naikrovekawesoem
20:46.26[TK]D-FenderKobaz: Yes, I have seen some retarded PRI's do this...
20:46.29Naikrovekawsum
20:46.36dustybini now have 2 lines with the same SIP because of the webgui config
20:46.51Naikrovekdustybin: yeah you can handle two calls on each line as well
20:46.55Naikrovekpolycoms are awesome
20:47.10[TK]D-Fenderdustybin: delete the "-phone" config and pull the power on the phone hard.
20:47.24[TK]D-Fenderdustybin: that should kill your overrides.
20:47.26Kobaz[TK]D-Fender: but i think i'm having the same type of issue... when callerid is empty... i get a reset of callerid name in from the pri *after* i set callerid(name)
20:47.35Naikrovekis away
20:47.42Naikrovekfake status update
20:47.43Kobazthat's my assumption... i'll need to turn on pri debug
20:47.47[TK]D-FenderKobaz: How much time does your call take to go through processing before that set?
20:47.56dustybin[TK]D-Fender: ok!
20:48.25doolittleworkwcselby: still there?
20:48.31wcselbyfor now
20:48.46Kobaz[TK]D-Fender: shouldn't be more than a hundred ms
20:48.54Kobaz[TK]D-Fender: it runs by pretty quick
20:49.05[TK]D-FenderKobaz: Time for some artificial delay..
20:49.07doolittleworkk working fine if the connected calls is Sip, overflows and plays the message
20:49.20Kobaz[TK]D-Fender: i already have a .5 sec delay on initial call pickup
20:49.21*** join/#asterisk scalex000 (n=chatzill@190.80.201.96)
20:49.24Kobazdo i need more?
20:49.27doolittleworkwcselby: for some reason it does now wanna work with zap
20:49.30scalex000hello
20:49.38[TK]D-FenderKobaz: evidence seems to say "yes"
20:49.42Kobazheh
20:49.47Kobazi even tried 1.5sec
20:49.49Kobazi'll try two
20:49.58Kobazoh... i know
20:50.06Kobazif the callerid is empty... i'll wait 2 more seconds
20:50.10Kobazotherwise it works fine
20:50.13wcselbydoolittlework - okay.......?
20:50.16scalex000I need to ask something about how polycom sip work and how to make dialplan.
20:50.20doolittleworkwcselby: does the g in the dial command only work with sip2sip calls?
20:50.29p3nguinnaikrovek, wcselby: Cisco says, "Configuration files reside in a TFTP server subdirectory (you can specify the location of this subdirectory with the tftp_cfg_dir parameter)."  Now I just need to figure ou where that parameter belongs.  :)
20:51.05wcselbydoolittlework - where have you shown me anything about the call that's not working?
20:51.23[TK]D-Fenderwcselby: You learn quick young Padawan...
20:51.28wcselbydoolittlework - as far as I know the g option should work on Zap / dahdi calls
20:51.42[TK]D-Fenderit works REGARDLESS of the called channel
20:51.48wcselby[TK]D-Fender - the more I help people, the more I understand your pain
20:51.54wcselby:)
20:52.05[TK]D-Fenderis a well of infinite sorrow...
20:53.19wcselbydoolittlework - until I see what's actually happening, I can't really suggest anything.  please paste the cli output (with verbose set to 30 or higher) of a call that's not doing what you think it should be doing.
20:54.04doolittleworki am batling to get the zap card to work for outbound
20:54.25wcselbydoolittlework - resolve your underlying issues before you try to resolve these easier issues
20:54.42wcselbydodgy network card, zap card not doing what you want, etc
20:56.33doolittleworkhey i am new to linux, i am trying here, big learning curve if one has been stuck betwwen windows and gates
20:57.07wcselbymay I suggest a new keyword for infobot - ~useful - Useful information to pastebin to help troubleshoot your issues - cli output (with verbose set to 30 or higher); relevant sections of .conf files; relevant DEBUG (SIP, ZAP, etc) info.  That'd be a good start anyways
20:57.48wcselbydoolittlework - I understand, but it's difficult to tell if the issues you're having are due to the underlying issues you've mentioned or if they're associated with asterisk
20:58.05wcselbydoolittlework - especially when you won't paste any of the requested information
20:58.12[TK]D-Fenderhttp://punditkitchen.files.wordpress.com/2009/02/political-pictures-bill-gates-campaign-fail.jpg
20:58.22*** join/#asterisk trebaum (n=trebaum@ip68-8-175-208.sd.sd.cox.net)
20:58.48[TK]D-Fenderwcselby: I have on for that....
20:58.53[TK]D-Fenderone*
20:59.09wcselby[TK]D-Fender - share please
20:59.10[TK]D-Fender~wmmfpb
20:59.11infobot[~wmmfpb] WHERE'S MY MOTHER-^%#ING PASTEBIN?!?
20:59.12Kobazokay
20:59.19wcselbyhahahahaha
20:59.19Kobaz[TK]D-Fender: i got a culprit call
20:59.20Kobaz[TK]D-Fender: http://pastebin.ca/1568877
20:59.20doolittleworki know i will get my facts straight then come back, if i set the cli to 30 there is a huge load of data to type ,so i will get network sorted and come back, thx for listening and advise
20:59.32wcselbynp
20:59.37wcselbyi'll be back tomorrow, probably
20:59.39Kobaz[TK]D-Fender: no callerid... and i SET callerid(name)... and then on the phone it's unknown
20:59.50*** join/#asterisk ibercom (i=d9d85043@gateway/web/freenode/x-szjxljykkvapmpyp)
21:00.06wcselbytime to hit the road, night all
21:00.17Kobaz[Sep 16 16:52:47] VERBOSE[20007] logger.c:     -- Executing [s@handleIncomingCall:30] NoOp("DAHDI/1-1", "DEBUG: 1253134364.11925 Callerid Name: TEST Aseracare") in new stack
21:00.29Kobazthat's my noop of the callerid(name)
21:00.34Kobazbefore i send it to the dialer
21:01.31trebaumDoes anyone here have any experience working with Colt Telecom in the NL?
21:01.52[TK]D-FenderKobaz: I don't see the call that goes out after that
21:02.32Kobaz[TK]D-Fender: i got the middle of a previous call in there...
21:03.00dustybinever since i deleted all the information on the webgui, i have lost my extensions!!
21:03.06[TK]D-FenderKobaz: Never any complete evidence...
21:03.21[TK]D-Fenderdustybin: Excellent, now you can do them right...
21:04.02dustybinyes, they were in the wrong place
21:04.03Kobaz[TK]D-Fender: heh... it's really hard to get a nice debug here... there's a bazillion calls coming in at the same time
21:04.10dustybini just created a dir called polycom
21:04.15dustybininside that is the .cfg
21:04.23dustybinand other dirs, like contacts, logs, overrides
21:04.45*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
21:04.48*** mode/#asterisk [+o Deeewayne] by ChanServ
21:06.00doolittleworkcan one record 24 fxo on a tdm24xxp card using GenuineIntel Intel(R) Core(TM)2 CPU 6700 @ 2.66GHz with 2 gig memory?
21:06.40[TK]D-Fenderdoolittlework: Sure.
21:07.05doolittlework[TK]D-Fender will the memory be enough?
21:07.17[TK]D-Fenderdoolittlework: * doesn't save to RAM you know..
21:07.54doolittlework[TK]D-Fender: but surely takes up memory and cpu power to record 24 sim calls
21:08.57[TK]D-Fenderdoolittlework: And we had * & T1 card before the P4 was out...
21:09.21doolittleworkwhat spec was your pc [TK]D-Fender?
21:09.47doolittleworkt1 30 or 20 channels
21:09.53[TK]D-Fenderdoolittlework: Its not the spec of the server, its the bandwidth of your pipe :p
21:09.55doolittleworkuk or brit?
21:10.02[TK]D-FenderUK doesn't have T1
21:10.02doolittleworkusa or brit?
21:10.11Kobaz[TK]D-Fender: heh... debugging output from asterisk is so hard to work with... theres no unique identifier for each call for non-dialplan output
21:10.12[TK]D-Fenderdoolittlework: C) None of the above
21:10.13doolittleworkok e1 right
21:10.25doolittleworkwhere u from [TK]D-Fender?
21:12.41ayesoHas anyone here ever reported a bug to the development team? Were they responsive?
21:12.56ibercomVoicemail with IMAP need a lot of connections to imap server. Is it possible minimize the connections number ?
21:13.42ibercomI have 500 extensions/voicemail users.
21:13.54doolittleworkcan one use the Chanspy command to spy on a specific sip extension?
21:16.10[TK]D-Fenderdoolittlework: ChanSpy spies on a CHANNEL
21:17.32doolittleworkwill extenspy work?
21:18.31*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
21:18.32ayesoAnyone know anything about the memory leak in the current version of comedian mail?
21:18.58grandpapadotayeso: svn?
21:19.27ayesoAsterisk 1.6.1.1
21:22.33ayesoIt leaks when calling functions to leave messages, Iv run it through ValGrind, but still analyzing the output.... i have to reboot * every night because of it.
21:23.04dustybinim unsure what this file does: 0004f2251870-phone.cfg  my log is trying to find it
21:23.54ayesoThis is using local storage, I have not tried IMAP
21:25.15[TK]D-Fenderdustybin: that would hold settings manually overridden diectly in the phone
21:25.18*** join/#asterisk ZX81 (n=ZX81@121.74.228.111)
21:25.40[TK]D-Fenderayeso: 1.6.1.6 = current
21:26.02[TK]D-Fenderayeso: When you're 5 vers behind, you know what the automatic answer is
21:26.09ayeso[TK]D-Fender: Grinded 1.6.1.6 as well same issue
21:26.12[TK]D-Fenderdoolittlework: both will work.
21:27.13ayeso[TK]D-Fender: I haven't diffed app_voicemail.c in 1.6.1.6 and 1.6.1.1 but I'm willing to bet there is no change.
21:28.30ZX81Was there a fire at Digium?
21:28.59dustybinZX81: VIC 20
21:29.06ZX81:)
21:29.33*** join/#asterisk seanmh (n=johndoe@c-69-254-131-168.hsd1.nm.comcast.net)
21:29.55dustybinthere seems to be a problem with sip communication to asterisk
21:30.09dustybinwhen the phone boots, asterisk doesnt say anything at all
21:30.12dustybinand i have no extensions
21:30.13ayesodustybin: sip set debug on
21:30.29dustybinayeso: asterisk isnt the problem
21:30.36dustybinit was working ok before
21:30.54[TK]D-Fenderdustybin: Because we flushed your reg info <-
21:30.58dustybinthere is some kind of problem with either, server.cfg
21:31.05[TK]D-Fenderdustybin: as I said, its time to do it RIGHT now.
21:31.27dustybin[TK]D-Fender: i have setup a reg.cfg file with my SIP auth settings
21:31.39dustybinbut the polycom isnt reading it or using it
21:31.43[TK]D-Fenderdustybin: where do I see that that file matters?
21:32.00dustybini will paste my configs
21:33.25dustybinhttp://paste.debian.net/46718/plain/46718
21:35.27dustybinhttp://paste.debian.net/46719/plain/46719
21:37.16dustybinhttp://paste.debian.net/46720/plain/46720
21:37.21scalex000hello
21:37.22dustybinthose are the 3 important one
21:37.51scalex000TK: Can I use ignorepat with polycom sip phone?
21:39.10Kobaz[TK]D-Fender:  http://pastebin.ca/1568921 finnaly got a clean one
21:39.12*** join/#asterisk blkry (n=chatzill@64.147.222.130)
21:39.17[TK]D-Fenderscalex000: No, only Zaptel/DAHDI FXS
21:39.21Kobaz[TK]D-Fender: no callerid... i set callerid name,.. and it's unkown kn the phone
21:39.47scalex000ok
21:39.53scalex000thanks
21:40.27Kobaz[TK]D-Fender: my debugging shows callerid(name) is what it should be until the end
21:40.37scalex000Tk: do you know how to create password before dial international call, any examples
21:40.58[TK]D-Fenderscalex000: "core show application read" , "core show application authenticate"
21:41.09scalex000ok
21:42.06Kobaz[TK]D-Fender: should i answer the call as late as possible?
21:42.20Kobazright now i do an answer, and continue to do some processing
21:42.41Kobazsometimes i need to play tracks and stuff before sending it to a phone. so i do an answer... but now that i think about it... i can do it at the track spot
21:43.14[TK]D-FenderKobaz: I'd delay it 2 sec to start
21:43.33Kobaz[TK]D-Fender: 2... okay
21:44.44[TK]D-Fenderdustybin: I recommend you trash those configs and start based on the ones int he provisioning pack
21:45.09dustybinok
21:46.13dustybingoes back to the drawing board, this requires a fresh cup of tea
21:47.19dustybin[TK]D-Fender: the zip file didnt have a example server.cfg
21:47.34[TK]D-Fenderdustybin: that is not a standard file
21:47.39dustybinok
21:48.04Kobaz<PROTECTED>
21:48.07Kobaz<PROTECTED>
21:48.10Kobazsame problem
21:48.17Kobaz[TK]D-Fender: is the provider killing me?
21:48.32Kobazor is this an asterisk/config issue
21:48.41[TK]D-FenderKobaz: as a test doa  long series of consecutive wait/Noop's to conunt the arrivat
21:48.43[TK]D-Fenderl
21:49.04Kobazcocout?
21:49.16Kobazcoconut
21:49.30[TK]D-Fendercount
21:49.41*** join/#asterisk xpot-mobile (n=james@173.8.94.1)
21:50.44Kobazi wait 8 seconds.. same problem
21:50.45Kobazheh
21:51.25Kobazi have some test did's.. i can do some simple tests
21:51.36[TK]D-FenderKobaz: Time when it DOES come in.  relative to entry / answer
21:52.17Kobazk
21:54.51*** part/#asterisk ZX81 (n=ZX81@121.74.228.111)
21:56.05*** join/#asterisk MaliutaLap (i=nikolai@203.39.87.98)
21:57.02*** join/#asterisk jaytee (n=jaytee@unaffiliated/jaytee)
21:57.17*** part/#asterisk ibercom (i=d9d85043@gateway/web/freenode/x-szjxljykkvapmpyp)
22:00.19dustybinphone1.cfg is a horrible file to edit
22:00.26dustybineverything is one big wrap
22:00.45p3nguineh... is it supposed to be that way?
22:00.48[TK]D-Fenderdustybin: don't like wrap then :)
22:00.53[TK]D-Fenderp3nguin: Yes
22:01.11MaliutaLapmo'ning Mr 'Fender
22:01.39dustybinat last!! i have communication!!!
22:01.41doolittleworkwhat is the variable for noop''ing the sip extension number that inisiates the call?
22:02.10[TK]D-Fenderdoolittlework: You can already see that in CLI just by the channel
22:02.21dustybinmy phone says 'John Doe'
22:02.23dustybinwho the hell is that
22:02.30doolittleworki wanna use the variable in the dialplan
22:02.42[TK]D-Fenderdoolittlework: ${CHANNEL}
22:02.53MaliutaLapdustybin: you've been haxor3D! ;P
22:03.23MaliutaLapdustybin: probably has something to do with how you configured the SIP, "John Doe" is the example name used in thebook
22:03.26doolittlework[TK]D-Fender: do you need to set it up in your sip.conf?
22:03.37*** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd)
22:03.54[TK]D-Fenderdoolittlework: ${CHANNEL} <-- this is the CHANNEL that is created based on the peer creating the call
22:05.05doolittleworkthanks
22:06.27Kobazokay
22:06.28Kobazwell
22:06.33Kobazi simplified everything
22:06.38Kobazthis is so fscked up
22:06.55Kobazhttp://pastebin.ca/1568952
22:07.41Kobazhttp://pastebin.ca/1568954 dialplan
22:08.50*** join/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc)
22:08.52Kobazso umm
22:09.01Kobazwhat could possibly be the problem
22:09.08mercutiovizDude, did I read that tweet correctly? Was there a fire at digium hq?!
22:10.20Chainsawmercutioviz: There was, and everyone is okay.
22:10.39mercutiovizokay, good
22:10.46mercutioviznever been there, never seen it, just a pic
22:10.53Chainsawmercutioviz: This has turned into an impromptu news article: http://www.venturevoip.com/news.php?rssid=2238
22:11.29doolittleworkwhat am i doing wrong with this variable Set(CALLFILE = ${AGENTINFO}-${TIMESTAMP}-${UNIQUEID}), i only get Set("SIP/1000-0a0f3e70", "CALLFILE = --1253138940.31"  only the unique id works
22:13.10*** part/#asterisk mercutioviz (n=michaelc@freeswitch/developer/msc)
22:20.12Kobazdo de do
22:21.03[TK]D-Fenderdoolittlework: First I don't know about, second - "core show function CHANNEL"
22:21.16Kobazheh so
22:21.18Kobaz[TK]D-Fender: any idea?
22:21.31[TK]D-FenderKobaz: Dunno
22:21.32Kobazi hate to be pestering everyone.. but... heh... this is kinda bad
22:27.25MaliutaLapKobaz: have I missed something, it seems to be doing what you want
22:28.15KobazMaliutaLap: the phone gets 'Unknown' as the callerid
22:28.20Kobazif there is no callerid on the incoming call
22:28.24Kobazit's not what i want
22:28.46Kobazit's like... if there is no callerid on the incoming call... the callerid is locked
22:29.42MaliutaLapKobaz: and you don't have "asrecieved" in the dahdi conf anywhere?
22:30.25Kobaznope
22:32.00*** join/#asterisk ebroad (n=EB@72.11.213.195)
22:32.25ebroadi need a recommendation
22:32.42ebroadim looking for a good cheap hard phone that supports video
22:32.55ebroad*hardphone
22:33.05ChainsawGood, cheap, video support. Pick any two.
22:33.06*** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net)
22:33.20ebroadhehe
22:33.37ebroadlets try cheap and video support
22:33.39MaliutaLap's true
22:34.06Chainsawebroad: I'm sure GrandStream will do something resembling a videophone.
22:34.28ChainsawGXV-3000 or so.
22:35.14ebroadlooks nice
22:35.21ebroadanybody using one?
22:35.37ChainsawSure, on your 1970s retro Ikea desk.
22:35.47*** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk)
22:36.03ebroadeh, this is for home
22:36.30ebroadi wouldn't roll these out in the enterprise
22:36.59mmlj4smart man
22:37.54doolittlework[TK]D-Fender i am lost again how do i datestamp a monitored file?
22:38.37[TK]D-Fenderdoolittlework: "core show function STRFTIME"
22:39.00doolittlework[TK]D-Fender: is that not to set your system time?
22:39.48[TK]D-Fenderdoolittlework: "core show function STRFTIME" <------
22:45.31*** join/#asterisk pthreadd (n=thread@85.138.26.234)
22:46.31pthreaddhello everyone
22:47.14pthreaddim having a problem making a zap trunk to work with a te210p dual-span card
22:47.22pthreaddi can receive calls
22:47.38pthreaddbut im unable to make calls through any of the PRI links
22:47.55*** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
22:48.35pthreaddand all i can see in documentations is just create a zap trunk with zap channel nr etc etc and everything will be fine
22:49.04pthreaddbut it isn't working anyway
22:49.20pthreaddif any of you can point me some usefull documentation about this topic
22:49.41[TK]D-Fenderpthreadd: Show us your configs and call attempt
22:49.42[TK]D-Fender~pb
22:49.43infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
22:49.44[TK]D-Fender^^^^^^^
22:50.21pthreadd[TK]D-Fender zaptel and zapata?
22:50.42[TK]D-Fenderpthreadd: Yes, and CLI output of the failed attempt
22:50.47pthreaddok
22:50.51pthreaddjust a sec
22:51.04pthreaddill give the link in a minute
22:51.20doolittleworkthx [TK]D-Fender: learned something new
22:53.24el_critterHi, how can I use dahdi_monitor to determine my PSTN busy pattern?
22:57.10*** join/#asterisk Toerkeium (i=Toerkeiu@201.216.206.221)
22:57.38*** join/#asterisk garymc (n=garymc@host86-162-159-166.range86-162.btcentralplus.com)
22:58.25garymc[TK]D-Fender Managed to open them ports on my home router....... it works the bomb now ;)
22:59.16[TK]D-Fendergarymc: And I was sure you'd need this beaten into your head a few hundred thousand more times...
22:59.41garymcYou where only right :S
22:59.54garymcI got a few more beatings in me yet ;)
23:00.03garymcbefore i go down
23:00.06garymc:S
23:00.47garymcWell just wanted to come in and thank you...... Thank you very much for the help :)
23:01.57garymcGood night
23:02.01*** part/#asterisk garymc (n=garymc@host86-162-159-166.range86-162.btcentralplus.com)
23:02.04[TK]D-Fendergarymc: Good, now hopefully you can move on to newer and more interesting things to break
23:02.10pthreadd[TK]D-Fender http://pastebin.com/d4a72354b
23:02.18pthreaddonly one PRI is configured at the moment
23:03.19[TK]D-Fenderpthreadd: And you should probably be dialing a GROUP, and not channel 1 fixed directly
23:03.45[TK]D-Fenderpthreadd: "G1" instead of 1
23:03.49[TK]D-Fender"1"
23:03.51pthreaddi've tried it
23:03.51*** join/#asterisk blackest_mamba (n=blackest@c-71-239-160-143.hsd1.il.comcast.net)
23:04.00pthreaddlet me try again
23:04.05pthreaddmaybe i've used g0
23:04.09pthreaddinstead of g1
23:04.12pthreaddsec
23:04.23[TK]D-Fenderpthreadd: correct this, then before your call also pastebin "zap show channels" zap show status" "pri show span 1"
23:04.38pthreaddok
23:07.34pthreaddhttp://pastebin.com/dd0278b5
23:07.37pthreaddthere it is
23:08.51[TK]D-Fenderpthreadd: looks pretty good. New call attempt please
23:09.07pthreaddk.. ill paste it in a minute
23:12.06*** join/#asterisk thegoat (n=jircii@c-71-224-180-83.hsd1.pa.comcast.net)
23:12.10*** join/#asterisk TimRiker (n=timr@bzflag/projectlead/TimRiker)
23:13.23pthreaddhttp://pastebin.com/mfb59716
23:13.42*** join/#asterisk raden (n=tanning@69-179-99-17.stat.centurytel.net)
23:14.01radenhttp://pastebin.com/m7424f693  <<< can someone tell me if this is vitelity or me
23:14.06radenim getting very annoyed
23:16.16[TK]D-Fenderpthreadd: "pri debug span 1" , "set verbose 10", try again
23:16.22pthreaddokey
23:16.25pthreaddsec
23:18.54raden[TK]D-Fender, can u traceroute inbound21.vitelity.net  or outbound.vitelity.net without error ?
23:19.17*** join/#asterisk propellerhead (n=yogurt2u@host251.200-82-124.telecom.net.ar)
23:20.26doolittleworkpthreadd: have u set your pridialplan and prilocaldialplan?
23:20.36doolittleworkalso try overlapdial=yes
23:21.41doolittleworkwhy would wanna do a traceroute IP if you working with isdn pri, or am i missing the plot?
23:22.49pthreaddhttp://pastebin.com/d6c57af0a
23:23.34[TK]D-Fenderpthreadd: Its not even trying and I don't see the debug I should be.
23:23.39[TK]D-Fenderpthreadd: restart * completely
23:23.42[TK]D-Fenderpthreadd: and retry
23:23.47pthreaddok
23:25.18pthreadddoolittlework i think pridialplan is in its default value.. national
23:25.37pthreaddor maybe not
23:25.44pthreaddits unknown
23:25.59pthreaddsec ill restart the asterisk and paste again the debug
23:27.47doolittlework[TK]D-Fender: can one save Mixmonitor files to custom directories in/var/spool/asterik/monitor folder?
23:28.47[TK]D-Fenderdoolittlework: You tell it where to save them..
23:30.40raden[TK]D-Fender, can u explain why with vitelity id get a temporary failure in name resalution at the hop nearest them ?
23:30.50[TK]D-Fenderraden: Nope.
23:30.52radenhttp://pastebin.com/m7424f693
23:31.18radeneven doing a route via ip i get the same issue
23:31.41radensomedays its there and somedays its not
23:32.05pthreadd[TK]D-Fender the output is similar to the previous one
23:32.14pthreaddi think it isn't dumping anything new
23:32.23pthreaddshould i increase verbosity?
23:33.51[TK]D-Fenderpthreadd: something is very wrong if it isn't trying to dial at all....
23:34.01[TK]D-Fenderpthreadd: And you've completely restarted *
23:34.06pthreaddyes
23:34.17pthreaddi dont understand this because i can receive calls
23:34.24pthreaddfrom the pri
23:34.42pthreaddi just can make calls
23:34.51pthreaddcan't*
23:35.14[TK]D-Fenderpthreadd: Show me an incoming call
23:35.22pthreaddok.. wait
23:38.32*** join/#asterisk coppice (n=chatzill@61.196.17.210.dyn.pacific.net.hk)
23:40.35bmoracaraden:  that name lookup failure is not for the second to last hop, it's FOR VITELITY, and all it means is that their PTR record isn't setup properly
23:40.53bmoracaraden: the second to last hop doesn't resolve because they probably disabled ICMP on it.
23:42.34pthreadd[TK]D-Fender http://pastebin.com/d7821f252
23:42.45p3nguinA trace to outbound.vitelity.net has every host replying, though.
23:42.57pthreaddcalling, ringing, hangup
23:43.05pthreaddworked perfectly
23:43.08*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:43.24pthreaddthere's no problems in inbound calls
23:43.33[TK]D-Fenderpthreadd: You are receiving as Zap and dialing as DAHDI.  Try dialing as Zap
23:43.45pthreaddhow do i do that?
23:43.56[TK]D-Fender~freepbx
23:43.57infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
23:43.57bmoracap3nguin: your route to their network may be different than his.  mine is also different.  i get there in 9 hops and all nodes respond as well.
23:43.59[TK]D-Fender^^^^^^^^^^^^^^^^^^
23:44.16pthreaddhummm k
23:44.22pthreaddthanks a lot for your time
23:44.28bmoracapthreadd: in amportal.conf, disable DAHDI compatibility mode
23:44.29pthreaddill try freepbx now :)
23:44.33p3nguinYeah.  It takes me 5 hops just to get out of my ISP.
23:44.40pthreaddbmoraca ok thanks
23:47.46*** part/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
23:48.43radenbmoraca, is that wy i would have registration issues ?
23:48.54bmoracaraden:  probably not, no.
23:48.59radenp3nguin, is that bad
23:49.13radenbmoraca, cause i only get that when i have registration issues
23:49.27p3nguinraden: No.  The route from me to vitelity is fine.
23:49.41radenhow about inbound20 and inbound 19 ?
23:51.07bmoracaraden: the only relevant part of your traceroute is the fact that you were able to get there with reasonable latency and that none of the hops between you and them had issues.  what are the problems you're aving?
23:52.02radencalls not going out or coming in
23:52.13radencallcentric been up weeks without issue
23:52.26radenvitelity everytime i turn around unreachable
23:52.27bmoracaare you registered with vitelity?  what does sip debug say?
23:52.42radenits working again out of the blue
23:52.50radennext time ill copy everything
23:53.08radenyeah i registered now but i lost registration for like 20 minutes 2 times today
23:53.14radencallcentric up all day
23:53.16bmoracarun sip debug next time it stops working.  that'll give a better picture of what's actually happening
23:53.19pthreaddthanks a lot guys
23:53.23pthreaddproblem solved
23:53.24el_critterhow can I use dahdi_monitor to determine my PSTN busy pattern?
23:53.25doolittlework[TK]D-Fender:
23:53.25radenbmoraca, thanks i will :)
23:53.35radeni have to go later guys
23:53.50pthreadd[TK]D-Fender special thanks to you.. you were a precious help :)
23:53.52bmoracahave fun
23:53.56pthreaddbmoraca thanks for the tip
23:54.38bmoracapthreadd: no problem...freepbx defaults that to ON in 2.5 and higher, i think
23:55.14doolittlework[TK]D-Fender:can one cp a file from one folder to another using an ivr, say for instance when they press one to cp a file from /var/spool/asterisk/monitor to /var/spool/asterisk/monitor/processed?
23:56.00bmoracadoolittlework: http://www.lmgtfy.com/?q=Asterisk+cmd+System
23:56.11doolittleworkthx
23:57.27*** join/#asterisk JayTee52 (n=jforde05@unaffiliated/jaytee)
23:59.51*** join/#asterisk wonderworld (n=w@ip-62-143-22-226.unitymediagroup.de)

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