IRC log for #asterisk on 20120810

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01:11.29fumonIs there anything short of a vpn that can do napt penetration for me?
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04:51.09*** join/#asterisk Dar1us (~doconnor@cain.gsoft.com.au)
04:51.51Dar1usI'm trying to write an asterisk extension which has >1 source file. Only OSX 10.8 (mountain lion) I get weird link errors (works on earlier OSX)
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04:58.28[TK]D-FenderDar1us: your wording is vague & incomplete
04:58.39[TK]D-FenderDar1us: please rephrase
05:02.34Dar1ushmm
05:03.09Dar1ushttp://pastebin.com/sx1HjgHt
05:03.26Dar1usboth chan_vomp.c and app_servaldna.c include asterisk headers
05:03.39Dar1usit seems that the latest xcode doesn't like that (see the pastebin)
05:04.02Dar1usI found that if I do this before including the headers it works..
05:04.06Dar1us#undef AST_API_MODULE
05:04.06Dar1us#define LOW_MEMORY
05:04.09Dar1uswell, it builds anyway :)
05:06.55*** part/#asterisk asteriskmonkey (~philip@74-51-38-204.telnetcommunications.com)
05:13.06Matt_Pis it the ring candice in indications.conf that asterisk uses to determine a pickup ?
05:14.19Matt_Pwhen i dont have call supervision
05:14.19Dar1usOK it loads OK
05:14.40Dar1usI am pretty surprised the default for AST_INLINE_API causes it to multiply implement each function (non-static)
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06:02.51v0lZymorning
06:05.21ChannelZhip hip hooray
06:21.08v0lZy;P
06:21.12v0lZyHi ChannelZ
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06:30.37kikohnlAloha!
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06:40.58v0lZyhi
06:41.49bulkorokhi
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06:49.39schmidtsgood morning
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06:50.39v0lZylo
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07:21.37CadeyANy guys in the UK online (in the southwest, west or central london area) ?
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08:09.21*** join/#asterisk Hisoka51 (~Hisoka51@79-201.104-92.cust.bluewin.ch)
08:09.50Hisoka51hi my friends
08:10.22Hisoka51i search one person for help in french ....
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08:27.57diabJe peux peut-etre pas t'aider Hisoka51 mais je peux traduire :)
08:29.39Hisoka51Merci c'est cool ^^
08:31.15Hisoka51Je vous explique en quelques mots, je suis un petit patron d'une entreprise en suisse et par challenge, j'aimerais installer un IVR sur nos lignes téléphoniques
08:31.30Hisoka51Pour ce faire j'ai installé TrixBox et une AVM fritz card.
08:31.31Hisoka51Le système en interne fonctionne parfaitement.
08:31.31Hisoka51J'arrive même à faire des appels depuis des clients X-lite vers l'externe ...
08:31.31Hisoka51Maintenant mon soucis sont les appels entrants.
08:31.31Hisoka51Pour vous expliquer ma configuration téléphonique. la carte Fritz est relié a mon boitier
08:31.51Hisoka51Quand je fais un appel sur un des 3 numéros de l'entreprise depuis un mobile on voit sur l'écran de mon serveur TrixBox une ligne qui s'affiche hdlc_down unknown prim(280) mais rien ne s'affiche sur mon client X-Lite alors que j'ai configurer le Inbound routes pour que sa arrive sur une extension d'un numéro existant sur le réseau.
08:31.51Hisoka51Je ne sais pas quels fichiers vous montrer afin de mieux comprendre mon soucis.
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08:39.46CadeyANy guys in the UK online (in the southwest, west or central london area) ?
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08:49.06WIMPyCadey: Do you need someone to come around?
08:49.48CadeyI need to speak with someone about a midium length conract
08:49.58Cadeycontract
08:50.08Cadeycant type today
08:50.40CadeyWIMPy: you in that area by any chance?
08:52.10WIMPyHisoka51: I can;t read french, but I think you've got troube to get a FritzCard working? These cards can only be used with the AVM CAPI. I'd recommend replacing it with a cheaper card that is
08:52.14WIMPysupported directely.
08:53.03Hisoka51wa
08:53.20Hisoka51What is the cheaper ?
08:53.50WIMPyCadey: Nope. To the north and quite a bit to the east across the water.
08:54.11WIMPyHisoka51: Anything with a HFC-S chip on it.
08:54.38CadeyCanada :) ?
08:54.43v0lZyWIMPy: hi
08:54.54Hisoka51I don't understand... =/
08:54.55Cadeyoh sorry east lol
08:55.09WIMPyThat includes HFC-USB, but that is not supported by dahdi, only misdn. See http://voice.yeti.dk/Asterisk_vs_ISDN/1
08:55.40CadeyNetherlands
08:55.49WIMPyCadey: Right at the border between DK and DE.
08:55.55Cadey:)
08:56.20Cadeygood asterisk guys are very hard to find in the UK
08:56.27v0lZyguys, can you help me out with blind xfer retrun
08:56.28Cadeyaspecialy ones that do contract work
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08:56.57v0lZyI'm blind transfering and if theres no answer after certain amount of time, i want to get  call back... how can I configure that?
08:57.15WIMPyv0lZy: Oh, that's very interesting...
08:57.45Cadeynot terably difficult
08:57.49WIMPyI came to the conclusion that it is only possible for some cases :-(
08:57.58Hisoka51I've installed MISDN and the card works to call for extern not working reception call
08:57.59Cadeythe use of the asterisk DB can solve that problem
08:59.13WIMPyHisoka51: Oh, they do support that card? Ok, so what's happening on incomming calls?
08:59.22v0lZyI suppose I use something like store variable of the user that got the call, then Dial(local/XX,20,to) .... and after that goto back to the user that picked up... but i have a transfer button... ,
08:59.34Cadeycorrect
08:59.49Cadeyuse the incomming calls caller ID as the helper
08:59.54Cadeysomthing like....
09:00.21CadeyInCall/{TheCallerID} = {Extension targeted}
09:00.30WIMPyYes. That's the point. You need unique caller IDs.
09:00.35Cadeythen when you forward the call or transfer the call, if the dial does not answer look up
09:00.40CadeyInCall/{TheCallerID} and redial
09:00.49WIMPyBut it becomes more interesting for attended transfers.
09:01.07v0lZyfor attended transfers, i dont need a call back
09:01.13v0lZythe person im transfering to either picks up
09:01.16v0lZyor they dont :D
09:01.29v0lZyif they dont
09:01.32v0lZyim the one hanging up
09:01.38Hisoka51When I get a call, the screen on the asterisk server I mark: hdlc_down unknown prim (280)
09:01.46v0lZyhowever, when i do that, that becomes a blind transfer... :|
09:01.53Cadeyv0lZy: do you have the ability to use the AMI?
09:02.04v0lZywhats that?
09:02.15Cadeyv0lZy: using that you can garantie the call back using the channel ID's and some code
09:02.26CadeyAsterisk Management interface
09:02.32WIMPyTroublw is that you might use an attended transfer to do a blind transfer.
09:02.53Cadeyhence the AMI :P
09:03.08Cadeywe can follow and track all the holds and joins
09:03.13Cadeywell, bridges
09:03.30WIMPyThe channel name is ok for SIP, but not for other channels.
09:03.32kaldemar${BLINDTRANSFER} might be helpful.
09:03.50Cadeyonly works if they use the asterisk featre code to transfer
09:04.02Cadeyif you use the phones buttons to Xfer or BXfer its not set :(
09:04.09Hisoka51WIMPy ?
09:04.15v0lZyWIMPy: thats exactly the thing... my phone when i click transfer just puts the call on hold and gives me another line. If i wait for the guy i call to pickup, then its attended transfer... if i drop before he picks up, then its a blind transfe.r
09:04.20WIMPyHisoka51: I assume that is a very old version of misdn?
09:04.51WIMPyv0lZy: That's the point. Technically it's still an attended transfer.
09:05.02Hisoka51WIMPy : How can I see the version?
09:05.08WIMPyI think it's sometime calld half blind or something.
09:05.29v0lZyWIMPy: i dont think it is. If i do proper attended transfers, callerid doesnt update. In all other situations, it updates.
09:05.54v0lZyno mater how long i wait, if i drop before they answer, they get the callerid, so i assume its blind xfer
09:05.59WIMPyHisoka51: That is actually not that easy as it is usually in the kernel, but maybe not. You are using chan_misdn?
09:06.52Hisoka51WIMPy, No, i don't use chan_misdn
09:06.57WIMPyv0lZy: That somehow works for attended transfers on current verisons.
09:07.32WIMPyHisoka51: chan_dahdi? chan_lcr?
09:07.58kaldemarCadey: it gets set.
09:08.09Hisoka51WIMPy : mISDN-mqueue ????
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09:08.52Cadeykaldemar: may be im thinking of ${ATTENDEDTRANSFER} that doesnt get set
09:09.02v0lZyWIMPy: with attended transfers i never get callerid of the original caller
09:09.13v0lZybut i do get it with blind transfers
09:09.29v0lZycan i control this somehow through dialplan or something
09:09.31v0lZythis transfer stuff
09:09.51Cadeyv0lZy: there isn another option
09:09.55WIMPyThe call back, not the callerid thing.
09:10.02Cadeyv0lZy: its not bullet profe but could help
09:10.10v0lZyim all ears
09:10.28Cadeyv0lZy: you will see in the CLI when you do a attended transfer the channel type is "local" I think
09:10.40Cadeyv0lZy: or am i thikning of forwarded calles from the phone :S
09:10.49Hisoka51WIMPy ?
09:11.15kaldemarCadey: that variable does not exist.
09:11.51WIMPyHisoka51: I just had to google for that one. I thought I had tried everything, but I hadn't heard of that one so far. Seems to be a special beronet version of the old mISDN (v1).
09:12.03Cadeykaldemar: humm well im almost 100% sure the variable only gets set if toy transfer a call using the feature code. If you use the handset the dialplan doesnt know the call has is being transfered
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09:12.51kaldemarCadey: chan_sip sets the variable. i checked the source.
09:13.03Hisoka51Have you Teamviewer ? please..
09:13.44Cadeykaldemar: If chan_sip sets the variable then it will only work if the handset sends the correct sip message to the server to make it aware of the transfer?
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09:15.37WIMPyHisoka51: I think if you want to use Trixbox, you'd better ask on #trixbox or some other support channel of theirs.
09:15.39kaldemarCadey: and that functionality is specified in the SIP rfc's. :P
09:16.39Hisoka51Thank's you, but I must to go... Sorry...
09:16.42WIMPyWhich one? And are there other RFCs specifying something else?
09:16.47Hisoka51Good bye !
09:16.52Hisoka51and thank's !!
09:17.06WIMPystill wonders what that mqueue thing is.
09:17.07Cadeykaldemar: yeah, the refer event
09:23.42v0lZyhm...
09:23.58v0lZyso... this callback thingy...
09:24.38v0lZyI mean... i suppose i can use dial with 20 seconds or something and then return to referer via variables
09:27.09v0lZyhm.. w8
09:27.13v0lZywhat context do i do this tstuff in
09:27.36v0lZyi have an internal context  for internal phones calling eachother
09:27.56v0lZywhen i do transfer...
09:28.10v0lZyif asterisk is unaware of this stuff... does it treat it as an internal to internal call?
09:28.47v0lZycause if thats the case, then i think ic an do something in my internal context
09:29.11WIMPyAsterisk doesn't know "internal".
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09:29.20WIMPyThe contexts are up to you.
09:29.31v0lZyagain, my poor expression
09:29.43v0lZyi put my phones into a context [internal] for example
09:30.11v0lZyif an internal phone calls an internal phone, thats the context where i handle things
09:30.27v0lZydoes that context apply when a user presses transfer' does that count as a call in the internal context?
09:30.46kaldemardon't think about "if an internal phone calls an internal phone", but rather "if an internal phone calls".
09:31.33v0lZybetter way of putting it yes
09:31.43v0lZytheres where i have patterns defined
09:32.40v0lZyis a transfer processed in the same way as a call? does it go through the context?
09:34.02kaldemarthe transfer is eventually a call.
09:34.03WIMPyAn attended transfer starts as a normal call.
09:34.23WIMPy+unrelated
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09:34.47v0lZyand unattended?
09:35.18WIMPyA blind transfer is an action on an existing call.
09:35.55v0lZyso, theres no context it goes through? it just shoots to whatever extension u send it to?
09:35.56WIMPyBut keep in mind that even if you use it as a blind transfer it may technically still be an attended transfer (terminated early).
09:36.33kaldemarthere's always a context.
09:36.35WIMPyI think the context still applies the same way.
09:36.53v0lZylet me code something
09:36.55v0lZyand paste
09:37.04WIMPyI.e. no difference at that point.
09:37.36kaldemarif you want to control the used context for the transfer, set __TRANSFER_CONTEXT to what you wish to use.
09:37.55v0lZyused context?
09:38.39WIMPyIsn;t TRANSFER_CONTEXT only for feature transfers?
09:38.41kaldemaryes. when a call lands in the dialplan, it always starts in a certain context.
09:38.51kaldemarWIMPy: no.
09:40.13WIMPyHmm. I guess I never manged to do a blind transfer from SIP then.
09:41.07v0lZymy environment: SIP/31 has context [SIP31] which has include => DialPatterns ...[DialPatterns] has a _XX,1,Gotosub(AppInternalCalls,InternalCalls,1(${EXTEN},...))
09:41.21v0lZywhen 31 dials 16
09:41.47v0lZy(16 has the context SIP/16 otherwise configured the same way)
09:42.01v0lZynow say that 16 picks up
09:42.05v0lZyand does transfer to 64
09:42.40v0lZythis should be happening in AppInternalCalls context
09:42.46v0lZyright?
09:43.22kaldemarwrong.
09:43.38kaldemarit starts in [SIP16].
09:44.01v0lZy[SIP16] only has
09:44.06v0lZyinclude => DialPatterns
09:44.14v0lZynothing else at this point
09:44.39v0lZyso what happens when 16 does xfer 64
09:44.55v0lZyit includes the dialpatterns definitions in [sip16]
09:45.15v0lZyand continues according to what it hits, right?
09:45.17v0lZybecause i have my voicemail there
09:45.26v0lZyand i can transfer to voicemail i think
09:45.51WIMPyIf you can call it, you can transfer a call to it.
09:46.13v0lZyhm... somethings not working at the moment though
09:46.23v0lZylet me recheck
09:47.03v0lZyah, think i found the proboem, 1 moment
09:55.56v0lZyyeah, i can transfer to voicemail
09:56.08v0lZylets see if i can do it blind too
10:02.24v0lZyok i think this is what im gonna do
10:02.38v0lZyfirst, ill put a noop somewhere to check something
10:05.00v0lZyok, yeah, its in the AppInternalCalls
10:05.13v0lZyi think i got all the pieces i need to resolve this
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10:07.58xoverukhi
10:08.16xoverukcould you advise me how I can trace a call in IAX? I wold like to determine if call transfer is working
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10:16.05v0lZywhat does sippeer return if it doesnt have information? as in the extension doesnt exist.
10:16.26v0lZyor wait, thats not a good idea..
10:17.00v0lZyi need to check if callerid(num) is an internal number or if its coming from the provider
10:17.23v0lZyah wait, brain knot, doesnt matter
10:24.16kaldemarusing SIPPEER with an extension is a bad idea since extensions are not peers.
10:27.49diabIs there a specific reason to limit the call pickup groups to 64 in asterisk?
10:28.21WIMPybits
10:29.24WIMPyBut there's a patch on the way to add named pickupgroups.
10:29.46diabnice
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10:33.00WIMPyYes, but I think I would have preferred multiple PICKUPMARKs.
10:34.21giulianozzhi all ! I've got a strange issue with asterisk 1.8.15. I have upgraded from 1.8.7 (src compile on centos 5.4 32bit). When I start asterisk or do a dialplan reload it takes about 30-40 seconds to complete the process, while with the old version it was about 1-2 seconds. The CLI output is also very slow compared to 1.8.7. Any idea on how to proceed ? cpu load os 0%, 2gb of total ram. Old version was also compiled f
10:35.25diabI have reached the 64 limits and i still have users asking for pickup groups, I think they mainly ask for that because of our phones which have only 5 ringtones (and all similars) and of course we can't add new ringtones...
10:36.14v0lZyshit, this is more complicated than i thought... this callback thing
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10:37.13Dar1usgiulianozz: your message was cut off after "Old version was..."
10:37.32v0lZyafter i dial something.... i need to check if theres a returnto variable and dial that or something... but when would i set this variable
10:37.41*** part/#asterisk Matt_P (~Parlane@ec2-23-21-49-97.compute-1.amazonaws.com)
10:37.42v0lZyi dont want to get called back every time i hangup a phone.
10:38.19WIMPy:-)
10:39.34v0lZythis one's a bitch.
10:40.27v0lZyseems u do it with TRANSFER_CONTEXT
10:40.34giulianozzOld version was also compiled from source. TIA !
10:40.37giulianozz<PROTECTED>
10:42.38v0lZyhm
10:42.44v0lZycan i use DIALSTATUS or somethign?
10:50.19v0lZydo transfered calls have any variables
10:50.20v0lZyor something
10:50.28v0lZyso that u can tell if the call was a transfered one or not?
10:51.03WIMPykaldemar told you about BLINDTRANSFER and ATTENDEDTRANSFER.
10:52.18v0lZyWIMPY: yeah i dont want to diferentiate that at this point yet
10:52.22v0lZybut i think i found something on google
10:52.28v0lZyTRANSFER_CONTEXT which kaldemar mentioned
10:52.39v0lZyapparently all calls that are transfered go through that context, right?
10:53.24WIMPyOnly blind transfers for obvious reasons.
10:53.36v0lZywhere do i set this?
10:53.44v0lZy[globals] context?
10:53.55WIMPyOn the transferring channel.
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11:20.52Dar1usI'm trying to use sqlite via odbc with openbts and I get "res_odbc.c:1390 _ast_odbc_request_obj2: SetConnectAttr (Autocommit) returned an error: HY009: [SQLite]invalid length or pointer"
11:20.58Dar1usstuff still seems to work but it's annoying..
11:22.39xoverukany ideas why after my iax2 call transfer I loose incoming voice
11:22.47xoverukbut my receiver can hear me perfectly.
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11:25.54oyugikHey guys does anyone know any Call Center software that intergrates with asterisk
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11:37.17v0lZyWIMPy: if im using TRANSFER_CONTEXT
11:37.21v0lZywhen does it get applied?
11:38.43WIMPyWhen you do a blind transfer.
11:39.55WIMPyAn attended transfer is just a new and unrelated call until the transfer completes.
11:43.10CboxHi,
11:45.15xoverukWIMPy: After the xfer the user can hear me, but I cannot hear them
11:45.20xoverukI can see traffic point to point
11:45.30xoverukwhy would I lose incoming voice?
11:46.35WIMPyxoveruk: Give us the whole picture. What devices are involved?
11:52.27xoverukback in one hour, if you are arond I would appreciate your help then,
11:52.29xoverukthank you.
11:52.30*** part/#asterisk xoveruk (~rich@193.220.55.57)
11:53.48v0lZyWIMPy: just noticed somethign: if i do transfer.. the music on hold stops, but theres no ringing
11:54.17v0lZythe phone im transfering to rings, but the call thats being transfered isnt giving an indication that the other side is ringing...
11:54.21v0lZyand theres no more music on hold
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11:55.29CboxWhen i call a mobile phone with a callfile put in the spool folder and then call another mobile from the dialplan, i only get one cdr entry, but i have two calls. How can this be fixed?
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12:01.35kaldemarCbox: you have one call, but two channels. if you want another CDR entry, use ForkCDR in dialplan.
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12:03.29Cboxkaldemar i just tried to call ForkCDR before det second call and i get two cdr entrys. However, the billsec is not correct on any of the calls for some reason. And has the same value.
12:05.24kaldemarCombine ForkCDR with ResetCDR.
12:05.50Cboxkaldemar ok. will give that a hot. :) thx
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12:12.15Cboxkaldemar i still get the same billsec on both cdr's. Do you know if there's a flag i should use?
12:15.24WIMPyv0lZy: I get ringing.
12:17.01kaldemarCbox: ResetCDR should set billsec to 0 when executed.
12:17.29kaldemarCbox: what are you expecting to be correct and what are the values?
12:22.34Cboxkaldemar it seems that the first cdr has the billsec value of the second call. The duration is correct, but not the billsec.. e.g. Call A: duration: 22, Billsec: 6 and Call B: Duration: 15, Billsec: 6 I would expect Billsec of Call A to be 17
12:23.31kaldemarCbox: why? nothing has answered the call really.
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12:24.58Cboxkaldemar, that's true. But i need the values to be as close as the ones i get from the provider. Could it be solved by simply putting an Answer() in top of the dialplan you think?
12:27.11v0lZyWIMPy: any idea why iw ouldnt get ringing? Answer() or smoething like that ?
12:27.30v0lZyor is this a music on hold configuration..
12:28.40WIMPyThat's a channel thing.
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12:31.06kaldemarCbox: should be. try it.
12:34.47Cboxkaldemar same result. :( But now i have an idea what's going on. Perhaps it will work if i dial a local channel and have that answer the call. Might be worth a shot. :)
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12:39.12like_a_horsehi all. I'm using the ReceiveFax application and find sometimes it receives the fax 100% and other times not. Lately I see a fax is receives and written to file perfectly and then outputs an error: app_fax.c: FLOW T.38T FAX exchange complete and then just below it: app_fax.c: Transmission error. After some reading I found i need to add in a hangup section in my receivefax macro which i did but the call never reaches "exten -> h,1,..." . Any s
12:39.12like_a_horseuggestions?
12:40.22[TK]D-Fenderi it doesn't reach there then you've done something wrong in yor dialplan
12:40.44like_a_horsenote: the faxes do seem to be received perfectly by the ReceiveFax application - i have the .tif to prove it :)
12:41.04like_a_horse[TK]D-Fender, one sec, let me get the dialplan
12:43.55like_a_horsehttp://pastie.org/4449507
12:45.36like_a_horsethis part of my diaplan has changed many times but this the latest version i've been testing
12:50.09[TK]D-Fenderlike_a_horse, problem is the fact you're doing this in a MACRO
12:50.22[TK]D-Fenderthe rules for how it merges with the calling context is what screws you
12:50.43[TK]D-FenderMake it a regular context
12:50.53like_a_horseaha
12:51.10like_a_horsemakes sense as my out going faxes work 100% and they are not in a macro
12:51.15like_a_horsebut a regular context as u say
12:51.20like_a_horseok thanks
12:51.23like_a_horselet me give that a go
12:52.10v0lZyWIMPy: ringing is a channel thing?
12:52.45v0lZyi have no idea why i dont get a ringing on the callers side
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12:53.27xoverukhi
12:53.30WIMPyyes
12:54.14v0lZyis this because when my phone does blind transfer, it bugs out of the conversation leaving it to asterisk, and asterisk hasnt bridged anything yet, so its not sending a ringing to the caller
12:54.27xoverukI have connected an IAX phone to a remote server and enabled transfer=yes on the iax extension, when I place a call the call is transfered, but I am not able to hear the receiving party, yet they are hear my voice.
12:58.40WIMPyxoveruk: Give us the whole picture. What devices are involved?
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13:05.33v0lZyhttps://issues.asterisk.org/view.php?id=9650
13:05.48v0lZybut my version is 10.5.1
13:07.32WIMPyTo check with chan_sip I'd have to fetch a 2nd sip phone.
13:08.06xoverukWIMPy: Citel IAX phone connects to reote server, remote server passes call to another server, the remote server then passes the call to the iax2 so that they communicate directly.
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13:08.20xoveruklocal and remote phones are using NAT
13:08.55xoveruknotrafer=yes, no problems, notransfer=no, call transfers, I cannot hear remote user, but they can hear me.
13:09.32WIMPySo it's all IAX, bith btween the phones and the servers and between servers?
13:10.13WIMPyAnd with transfer do you mean transfer (by a person) or just connected?
13:11.03xoverukcorrect
13:11.16[TK]D-FenderIAX "transfer" = REINVITE
13:11.26xoverukthe IAX transfers the call PTP as the origin is a remote extension, no local server.
13:12.07xoverukIAX extension (WAN)<------>IAXserver(WAN)<---------->IAXserver(WAN)-IAXextensionlocal
13:12.11WIMPySmells like a firewall issue to me.
13:12.35xoverukI can see on the remote server that iax in both directions.
13:13.04WIMPyFor that to work you will need to forward the IAX port.
13:13.04xoverukbut here potentially I am not receibing the packets.
13:13.09xoverukNo way to check
13:13.18xoverukah
13:13.38xoverukOK I see
13:13.44xoverukI cannot do that as I am using two phones
13:13.54WIMPyAutomatic NAT rules only work for replies.
13:13.58xoverukso would have to use public IPS on them instead?
13:14.03WIMPyUse different ports then.
13:14.11xoverukyes, very good point.
13:14.32xoverukcannot do that, no access to the router.
13:14.54xoverukIs it safe to use WAN IPs on the IAX phones?
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13:15.23WIMPyDo you trust the phone?
13:15.36v0lZyWIMPy: when i dial, i use only ,to option
13:15.37xoverukI dont trust anything these days
13:15.38v0lZys
13:15.42v0lZywould i need anything else to get the ringing'
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13:15.56WIMPyThen don't connect to the internet.
13:16.08xoverukIt is a Citel C4110
13:16.11xoverukany good?
13:16.23xoverukunfortunately I need to make a living
13:16.24WIMPyNot head of that before.
13:16.48xoverukcheers anyway
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13:17.39WIMPydidn't know there were working IAX desktop phones.
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13:25.44jeffspeffso, having an issue with realtime IAX through mysql
13:25.47jeffspeffthe IAX peers don't show up when i do a iax2 show peers
13:25.50jeffspeffbut i can do a realtime load and it will show the info from the database
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13:27.18xoverukWIMPy: I am not quite understanding why when the call is transferred that the router cannot translate the IP address.
13:27.49jeffspeffxoveruk, do you have directmedia turned off?
13:28.00xoveruknot sure what that is
13:28.06jeffspeffusing sip?
13:28.10xoverukIAX
13:28.15jeffspeffoh
13:28.22jeffspeffnvm then
13:28.25WIMPyxoveruk: Because the reply comes from an unknown source.
13:29.13WIMPyBut you could just try another firewall that might work better.
13:29.34xoverukbecause the connection was not made to the destination IP originally?
13:29.47WIMPyyes
13:30.03WIMPyWell, it's not even a connection.
13:30.04xoverukso who starts the transfer connection?
13:30.17xoverukwhat do you call a UDP flow?
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13:30.42WIMPyNot sure, but from your observations it must be the destination.
13:31.07xoverukit appears that way, it is annoying that i cannot captiure the inbound traffic
13:31.11xoverukno access
13:32.51WIMPyActually I don;t even know if natted connections can be transfered by design.
13:33.15WIMPyMaybe someone more in to IAX can answer that.
13:33.49[TK]D-FenderIAX does not care about NAT.  It always goes to the sourcee IP&port
13:35.15WIMPyBut who tells the 3rd party where to connect? The originator or the "man in the middle"?
13:35.19jeffspeffhey [TK]D-Fender any experience with realtime IAX in mysql?
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13:36.18xoverukWe just tried the call in the other direction, the remote user called me, that person has a local iax server. This time I could hear them, but they could nto hear me.
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13:36.32WIMPystill thinks it's just a firewall thing, but sometimes it's a good idea to make sure it's the only possible source of the issue.
13:36.36xoverukthat does not make sense as that server receive all IAX traffic
13:37.06xoverukhmm
13:37.26WIMPySo you have transfers disabled on that server?
13:37.56xoverukmust I set it on each peer and each iax extension?
13:38.01xoverukI enabled it only on the extension.
13:38.08xoveruknot on either of the servers
13:38.18xoverukwhat is the default for iax trunks?
13:38.37WIMPyNo idea what the default is.
13:38.49WIMPyBut don't call peers extensions.
13:39.47xoveruki am using trixbox, when I connect a phone its called an extension
13:40.01xoverukwhen i create a link between two servers its listed as a trunk
13:40.12xoveruktrunk is just a waya of multiplex though isnt it ?
13:40.16xoverukso its misleading ?
13:40.19[TK]D-Fenderjeffspeff, Not really...
13:40.26jeffspeffok
13:41.08[TK]D-FenderNo, they use "trunk" as a resource to to call to terminate other calls.  So that it isn't expecting the reciever to be a singulr phone.
13:41.26[TK]D-FenderFreePBX  "extension" is meant to represent an entire user tha may have dedicated VM, etc
13:42.04xoverukmust is set notransfer=yes on the peers and the extension?
13:42.36xoverukdoes it even need to be set on the extension?
13:42.43xoverukmakes no sense to me as it is not a middle man
13:45.36like_a_horse[TK]D-Fender, Looks like the regular context did the trick. Thanks for adding on an extra couple hours onto my life
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13:48.52[TK]D-Fenderlike_a_horse, You're welcome
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13:51.15xoverukthis is annoying
13:51.33xoverukany iax gurus out there?
13:54.08oejIAX gurus are not that common? ;-)
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14:04.47WIMPyProbably not. It has been a little neglected anyway :-(
14:05.39xoverukdo people use SIP?
14:06.44WIMPyThe vice is everywhere.
14:07.04anonymouz666I use IAX2 heavily.
14:07.20anonymouz666because SIP sucks to communicate between two asterisk boxes.
14:07.56WIMPycertainly prefers IAX.
14:08.14[TK]D-FenderSIP works just fine with *.
14:09.08anonymouz666[TK]D-Fender: OK, try to have 2 boxes sharing the same config file (sip.conf with same user/peers) and Dial between them.
14:10.11VultureZanonymouz666, I have eben working OpenSIPs to accomplish probably the same thing
14:10.30anonymouz666I also work with SIP Proxy
14:10.53anonymouz666but not in all cases
14:10.53VultureZto filter the peers to the individual boxes?
14:11.09[TK]D-Fenderanonymouz666, Works fine....
14:11.41[TK]D-Fenderanonymouz666, The peer you'd use to pass from A to B can work just fine.
14:11.49anonymouz666[TK]D-Fender: no.
14:12.32[TK]D-Fenderanonymouz666, Yes.  I've clearly done this before.  While you may have failed you don't get to tell me that I did too.
14:12.35anonymouz666you auth let's say 1000 that is user and peer on machine A. in machine B with match user from and ask for auth AGAIN.
14:12.45anonymouz666I don't want to rewrite things.
14:12.48[TK]D-FenderFix your peer between your boxes
14:13.36anonymouz666I can't do that in my setup, just does not work that way
14:13.45[TK]D-FenderSame device names for your other phones is irrelevant,  Fix the ONE you use to call from server A to server B
14:14.01[TK]D-FenderCan't do it?  What.. you have no control over your configs?
14:14.51*** part/#asterisk xoveruk (~rich@193.220.55.57)
14:15.58anonymouz666imagine I have machine B and A. We got a call in machine B queue, 1000 got the call. the from is the ANI coming from pstn.
14:15.59anonymouz666so
14:16.03VultureZ[TK]D-Fender, I keep coming back to this channel year after year and you are always still here. I seem to recall you hanging out in this channels for what feels like 10 years ago. It was back when Asterisk was forking
14:16.15anonymouz666the transfer between two machines will work.
14:16.24[TK]D-FenderVultureZ, Yes, pre CallWeaver...
14:16.59anonymouz666in p2p communication
14:17.06[TK]D-FenderVultureZ, and "Pre-Whole-Numbered-versions"
14:17.14VultureZyea, you Qwell and file are very familiar names, don't see Ariel_ here anymore
14:17.21anonymouz666if I dial from 1000 to 1001, that will not work.
14:17.32anonymouz666in different machines
14:17.39anonymouz666unless I set a peer
14:17.44anonymouz666the rewrite the fromuser
14:17.58VultureZAh yes, the days where timing was a nightmare
14:18.07anonymouz666because in the other machine will match always the user using the From field
14:18.19anonymouz666I don't have to worry with that with IAX2
14:18.25[TK]D-Fenderanonymouz666, Yes, you aren't jsut going to do a direct SIP call without a peer between your servers.  * isn't a proxy.
14:18.26anonymouz666I just have the exactly same config file.
14:19.10anonymouz666[TK]D-Fender: this is possible. why not? with share the same database, the * will know the contact field.
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14:19.46[TK]D-FenderNormally not a heathy thing....
14:20.38anonymouz666that's a SIP feature, you know the contact user, you know where to send the INVITE
14:22.25[TK]D-FenderI suppose so... why is it you have 2 servers sharing like that again?
14:23.28anonymouz666imagine that there's a sip proxy doing a register load balancing before asterisk
14:23.37VultureZ[TK]D-Fender, what would you recommend for a LB setup? OpenSIPS -> Asterisk(s) w/Realtime
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14:28.31[TK]D-FenderVultureZ, Don't really do LB
14:29.35VultureZbunch of medium office installs with just failover?
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14:55.35danfromukHi everyone. Why would Playtones(busy) result in silence and give the following error WARNING[31667]: translate.c:202 framein: no samples for alawtolin
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15:06.16navaismoRemember that call monitor program, well in windows it had a bug on windows, fixed. You can download from here asterisktools.blogspot.com. You can use the translate toolbar for English if you want to read.
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16:14.34vlad_starkovQuestion: what is the better name for incoming call logic: "dialplan" OR "incoming call routing" OR "incoming call logic"?
16:15.07[TK]D-Fenderdialplan
16:15.18[TK]D-Fenderevery call processing is "inbound"
16:15.52vlad_starkovnot "incoming" but "inbound"? Am I correct?
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16:19.06vlad_starkov[TK]D-Fender: thanks
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17:04.50RobOakesMorning everyone. Is this a good place to ask user questions?
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17:06.12RobOakesI'm evaluating asterisk for use in my business (there are 5 of us). I just installed it from the Ubuntu repositories, and I'm trying to setup the first user.
17:06.41RobOakesI've created an entry in sip.conf, but whenever I try to connect to it, I get an error that says: No matching peer.
17:07.20RobOakes[100]
17:07.22RobOakestype=friend
17:07.24RobOakeshost=dynamic
17:07.26RobOakessecret=100
17:07.28RobOakescontext=users
17:07.30RobOakesmailbox=100@default
17:07.49RobOakesThe settings were from a tutorial I found online. Does anyone have any idea why I might be getting the authentication error?
17:08.59RobOakesI am using Empathy to connect.
17:09.01navaismodo you do the sip reload in the asterisk cli?
17:09.45RobOakesUhm ... no. I was just exiting and then restarting asterisk.
17:09.51RobOakesLet me try that.
17:10.10navaismowell if you restart asterisk that load again the sip conf,
17:10.27navaismobut isnt not necessary to restart asterisk
17:10.50navaismoonly sip reload when you modify sip.conf
17:10.56navaismo~book
17:10.56infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
17:11.03navaismoRobOakes, ^
17:11.05RobOakesReloading SIP worked. I was just able to get it to work.
17:11.20RobOakesThank you. Also, thank you for the link.
17:12.55navaismono problem
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17:32.20jeffspeffi'm having a hard time trying to get an IAX trunk to register between two boxes
17:33.13jeffspeffiax2 show registry says it was rejected, but i can't find anything saying what or why it's rejected
17:35.57*** join/#asterisk jdoe (jdoe@falseprophet.ca)
17:36.14jdoeis there a way to explicitly destroy a sip channel on 1.8?
17:36.27jdoe"channel request hangup" isn't killing it.
17:36.58*** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee)
17:37.09anonymouz666jdoe: this command used to work
17:37.48jdoeno doubt it still does, but the channels are stuck. I think it's ASTERISK-19455 since I've been a bit lazy about keeping up to date.
17:37.58jdoebut I can't restart right now.
17:38.18danfromukHi everyone. Why would Playtones(busy) result in silence and give the following error WARNING[31667]: translate.c:202 framein: no samples for alawtolin
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17:44.51jeffspeffdanfromuk, what codec are you using?
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17:58.24bobb_WUhey everyone
17:59.00bobb_WUquick question- i have a dial plan defined extension that says the extension and records a message, then calls back and plays the message using a script and the "h" extension
17:59.19bobb_WUthat has bled over to another context so now every call to this server runs that script
17:59.24bobb_WUis there a way to seperate it?
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18:09.42danfromukjeffspeff: I'm using alaw.
18:09.52danfromukSorry for the delay in replying.
18:10.02paulcbobb_WU: How has it "bled over" ?  Through some kind of include?
18:10.39bobb_WUi'll look
18:10.41paulcI'd put that recording extension and the "h" extension in a context of their own, just those two, then in my "main" dialplan, have an extension Goto that other context/extension to do the recording.
18:11.43bobb_WUah that's smart
18:11.56paulcdoffs hat - I try ;-)
18:11.58bobb_WUno there isn't an include to the seperated context.  i'll look at re working it
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18:15.36haroldpheya
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18:16.14lorsungcuhi
18:18.32*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
18:18.38haroldpI have an asterisk server that connect to two sip trunks (jargon?) one for home, and one for my office.  For some reason, I can only get incoming  calls on one of the lines now.  Both can make outgoing calls.  I *do* see the incoming call hitting asterisk when i monotor the console
18:19.14haroldpany pointers on debugging that?
18:19.51*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
18:19.52ravnxwhat does it say on the console when it hits it and doesnt work?
18:20.17haroldpabout six miles of sip noise that I don't understand :)
18:20.23lorsungcupastebin relevant extensions.conf, sip.conf, sip debug and a cli ouput
18:20.40haroldpI only get output when I sip set debug on
18:20.48lorsungcuthen paste bin that
18:21.38haroldphah
18:21.45haroldpwell, you fixed it.  thanks.
18:21.48lorsungcunp
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18:22.02haroldpNow I need to beat something.  With a stick.
18:22.41lorsungcuwhat'd you do wrong
18:22.53haroldpthrows his hands up
18:23.04haroldpit's a config that has been working for months
18:23.26haroldpI think maybe my router lost some config
18:23.29lorsungcuhere's what i know so far:
18:23.49lorsungcuyou can get calls through only one of your two sip providers
18:24.09lorsungcuno CLI output, only when sip debug is on do you see anytihng
18:24.47lorsungcutend not to think its your router
18:24.51lorsungcuif you are seeing sip traffic
18:25.03lorsungcupaste bin sip.conf
18:25.08rillianis opus support still scheduled for asterisk 11?
18:25.09haroldpall of a sudden, I stopped getting incoming calls.  I reset some router config.  tested.  didn't work.  tested again to provide you with some pastbinery. worked.
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18:25.55vlad_sta_Question: What is the most common case about faxes (I'm talking about virtual fax, if you still use it)? Only 1 fax-machine per whole company OR 1 fax-machine per department OR 1 fax-machine per employee?
18:26.00haroldpassumes this will never happen again, heh :)
18:26.08lorsungcuok fuck it
18:27.59haroldpthanks for your attention though.  My sysadmin sense tells me it will just break again unless I figure out for sure why it broke.  But my programmer sense tells me I'm gonna get flogged if I don't stop dicking around with the phones and get back to work :)
18:28.07ravnxvlad_sta_: it depends on the company. i've got customers where everyone has their own fax DID, and others that have a single and others that have a few department. i wouldnt say anything is common.. except FAXES CAN BURN IN HELL.
18:28.22haroldphah
18:28.38vlad_sta_FAXES CAN BURN IN HELL --- that's right ;-)
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18:28.47jeffspeffagrees
18:29.05ravnxi spend more time supporting faxes than any other issue ever. i hate faxes.
18:29.13vlad_sta_In Russia faxing is still very common
18:29.31lorsungcuas in the US :p
18:29.33Hisoka51/scid -at1 amsg
18:29.34ravnxits amazing how much people still use faxes in the US
18:29.47CrazyTux[m]Hey guys I'm having an issue where a call comes from asterisk -> goes out and comes back in (different ruri) (spiraling) and asterisk is basically not processing the new invite/ruri and the call fails?  Any thoughts?  I've seen there have been previous bugs opened in regard to this back in 2007 and pedantic=yes needs to be enabled (which it is) ?
18:30.04vlad_sta_Why people still using faxes?
18:30.18ravnxthey like to hear the turkey sounds
18:30.21lorsungcuvlad_sta_: has timestamp, is realtime, when it works, it _works_
18:30.35lorsungcubut timestamp is a big deal
18:30.38CrazyTux[m]Referencing Bug File : https://issues.asterisk.org/view.php?id=7403
18:31.27coppicepeople don't really have an efficient alternative to FAXes, because many whine but don't provide real solutions
18:31.35vlad_sta_In Russia (as I understand) it's that just because our legislation does not "understand" e-mail :-)
18:33.36ravnxplus people say "oh i never got the email", with faxes they can say "i have the timestamp from the confirmation you got it printed"
18:33.49lorsungcupretty much
18:34.16ravnxstill virtual fax sucks its never 100%
18:34.36lorsungcui have 0 issues using vitelity
18:34.48lorsungcuhave no reason to handle it myself
18:35.13lorsungcubut using http://flowroute.com, have almost no issues
18:35.22vlad_sta_In our court you can show fax message on paper sheet and they will accept it. But it could be very problematic to show email, and prove that it is original and not faked...
18:35.39ravnxi work at something similar to vitelity, so i have to handle it. i think vitelity uses PRI's we've moved all our DIDs to SIP and ever since, its been just silly.
18:36.16lorsungcuyeah vote;lity is all pri
18:36.39ravnxwhen we had all PRIs for fax, it was awesome, always worked.
18:36.51lorsungcuwho is your SIP with
18:37.06ravnxLevel3 and Windstream
18:37.21lorsungcuyou have issues with l3?
18:37.25ravnxeven using the paid for digium t38 modules
18:37.47ravnxyeah its not a lot, id say 5 out of 100 faxes fail
18:37.58lorsungcuhmm
18:38.14lorsungcuthink we'll be moving to l3 for lots of stuff within a year
18:38.18ravnxusually t38 negotiation errors
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18:38.57ravnxL3 has been really great, expensive to LNP in DIDs but great voice service.
18:39.12ravnxi think their Sonus switches kinda jack with the media for t38
18:39.36lorsungcuhmm
18:40.19coppiceif sonus can break something, they will. it seems to be company policy
18:40.59vlad_sta_Does anyone faced with situation when boss of company don't like to share incoming faxes with some of employees?
18:42.18ErectBananais the PAP2T a good ATA with asterisk? I found a couple unlocked ones on ebay I could buy
18:42.21ErectBananalinksys
18:43.00vlad_sta_ErectBanana: I use PAP2T and it's fine
18:43.22ErectBananavlad_sta_: can both analog ports be in used at the same time?
18:43.24ravnxErectBanana: yah i use them too, they work, i don't think it does t38 though, you'll need SPA2102 or so
18:43.26ErectBanana*in-use
18:44.18ErectBananaahh
18:44.25ErectBananaI think I need t38
18:44.44vlad_sta_ErectBanana: as I remember it has 2 separate SIP setups
18:44.50coppiceyou can use both ports for G.711, but bit for low bit rate codecs. the 2102 has the same issue. I don't know about the new models
18:45.00coppices/bit/not
18:45.16ravnxive got one of the black new ones deployed and havent heard complaints yet.
18:45.37coppicethe new ones have very buggy code
18:45.48vlad_sta_ErectBanana: on asterisk side I use SpanDSP based fax, everything works just fine
18:45.56ravnxi think its "SPA112"
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18:46.31ravnxcoppice: i noticed in chrome the web interface goes wacked with it sometimes. have to use IE
18:46.35ErectBananawell there is also the SPA122
18:46.38ErectBananawhich is newer
18:47.34coppiceravnx: it might be settling down with recent updates, but a couple of months ago the 112 and 122 were giving people trouble with all sorts of features
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18:48.10kresp0ErectBanana: be careful with "unlocked" PAP2T as they may brick if you do a factory reset
18:48.28ErectBananayeah I think I might go with the SPA2012 or the SPA122 now
18:48.33ErectBananasince I need T38
18:48.39ErectBananaI want one of the ports connected to a fax
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19:02.41ErectBananathanks everyone for the feedback
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19:07.04lorsungcuso hungover
19:07.42*** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
19:08.11WIMPyDebugging Asterisk all night or drinking?
19:08.45lorsungcuthe latter
19:09.26ravnxlorsungcu: im suffering today too.. thursday night drinking is no good for friday's workday
19:10.00lorsungcunot a good idea.
19:11.43ravnxyeah, i knew it when i said "sure just one more" that it was gonna be too much, woke up at 3am in cold sweats, but queasy all day
19:15.48CrazyTux[m][TK]D-Fender: got a quick second? this is driving me insane.
19:15.59CrazyTux[m]-- Now forwarding SIP/proxy-4c044f90 to 'Local/blacklist@default' (thanks to SIP/proxy-009b6a50)
19:15.59CrazyTux[m]== Everyone is busy/congested at this time (1:0/0/1)
19:16.05[TK]D-FenderI am your co-pilot on the way to crazy....
19:16.12CrazyTux[m]Why .... spiral request is 100% always resulting in this.
19:16.23[TK]D-FenderPBthe whole deal
19:16.24CrazyTux[m]Asterisk -> proxy -> Asterisk (ERROR)
19:16.31CrazyTux[m][TK]D-Fender: PB  ? lol
19:16.42ravnxput peanut butter on it
19:16.46CrazyTux[m]lmfao...
19:17.08ravnxhe means post the whole output to pastebin
19:17.25CrazyTux[m][TK]D-Fender: https://issues.asterisk.org/view.php?id=7403 I see this bug report, but actually don't think it is applicable now since the Local URI IS CORRECT! ahh..
19:18.01CrazyTux[m]That is literally essentially the entire outout/log
19:18.15CrazyTux[m]The onyl thing that happens before it is a DIAL() to the proxy and the proxy sends it BACK to asterisk with a new RURI
19:18.19CrazyTux[m]s/onyl/only/
19:18.21navaismoDoes Digium is planning to move DPMA to RealTime?
19:18.21jeffspeffi'm trying to set something up so that i can playback a sound through page()
19:18.55jeffspefflike when phone auto answers, instead of hearing the audio from my phone, they hear a sound file
19:19.30WIMPyCan't you use multicast?
19:19.30CrazyTux[m][TK]D-Fender: Asterisk 1.4.25.1
19:19.53jeffspeffWIMPy, i highly dislike multicast
19:20.08[TK]D-Fenderjeffspeff, So Originate a call and use Playback
19:20.28[TK]D-FenderCrazyTux[m], ANTEDILUVIAN
19:20.53jeffspeff[TK]D-Fender, is there not something special in the page() command that makes the phones auto answer?
19:21.02[TK]D-FenderNo
19:21.30[TK]D-FenderYOU make the part that dials out do what is requred to make that specific channel do so
19:21.45CrazyTux[m][TK]D-Fender: had to look that up in the dictionary ;)
19:21.57[TK]D-FenderIn the case of SIP phones that support it, you add headers befoer calling out to the,
19:23.16jeffspeff[TK]D-Fender, ok, got the headers part done, the phones auto answer
19:23.58jeffspeff[TK]D-Fender, here's what i have http://pastebin.com/iUtdTNLF
19:24.13jeffspeffbut the Playback isn't done until after the call ends
19:25.57[TK]D-Fenderjeffspeff, You don't seem to grasp the dialplan.....
19:26.17jeffspeffi just don't think i'm grasping this part
19:26.22[TK]D-Fenderjeffspeff, it is LINEAR.  You dial, and the only reason for the next step to happen is because the call FAILED
19:26.35[TK]D-FenderOriginate <----------------
19:26.48jeffspeffi understand the linear part
19:27.01[TK]D-FenderApparently not since you put a playback AFTER a dial
19:27.11jeffspeffwon't originate just make a call at a certain context,exten,priority?
19:27.32[TK]D-FenderThere are 2 parts to it.
19:27.42WIMPyYes. A call, i.e. 2 channels.
19:27.50[TK]D-FenderWhat you call (Channel:), and where you dump them.
19:27.53[TK]D-Fender2 channels
19:28.06CrazyTux[m][TK]D-Fender: http://pastebin.com/6ExQYUPk
19:28.10CrazyTux[m][TK]D-Fender: complete log
19:28.14[TK]D-FenderSo one side does the playback, the otehr does the Page()
19:29.25[TK]D-FenderCrazyTux[m], Well so far I see the attempted redirect and immediate termination... now show the dialaplan to back it...
19:30.41jeffspeff[TK]D-Fender, so you're suggesting that (with my extension being 1506)   channel originate SIP/1506 1@page-context        and that the page context plays back the file?
19:31.34[TK]D-FenderSomething like that....
19:31.43jeffspeffthat would require me to answer the phone and doesn't let me set the sip headers to enable auto answer
19:31.54[TK]D-Fenderthen you'd better set them first <-
19:32.09[TK]D-FenderSo dial a channel that lets you do this.
19:32.11CrazyTux[m][TK]D-Fender:
19:32.11CrazyTux[m]exten => blacklist,1,Playback(phone_number_blacklist)
19:32.11CrazyTux[m]exten => blacklist,n,Hangup
19:32.28[TK]D-FenderTime to review the full list of * Channel types.  There is essential learning here...
19:32.42CrazyTux[m][TK]D-Fender: hmmm me?
19:32.46[TK]D-FenderCrazyTux[m], PB.  No stupid little fragments... you know better
19:32.56CrazyTux[m][TK]D-Fender: that is *it* once it spirals back
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19:33.09[TK]D-FenderCrazyTux[m], I also don't see a SIP warning in there.
19:33.09[TK]D-Fender...
19:33.14[TK]D-FenderOr any debug at all
19:34.49CrazyTux[m][TK]D-Fender: not sure what else I can show you lol
19:35.05CrazyTux[m][TK]D-Fender: I could paste the original dial plan but I don't see the relevence... and would be extremely difficult to do so
19:35.59CrazyTux[m][TK]D-Fender: are you referencing a sip warning in the actual dial plan, I'm a bit confused here.
19:36.52WIMPyOr just use application playback as the second channel.
19:37.12CrazyTux[m]WIMPy: was that to me?
19:37.37WIMPySorry. No jeffspeff. I wasn't fully scrolled down.
19:39.10CrazyTux[m][TK]D-Fender: in all seriousness do you really need to see the earlier part of the dial plan?  If so I will extract it.. but it will be difficult
19:39.26CrazyTux[m][TK]D-Fender: this SIMPLE task is driving me insane that Asterisk is erroring out on a spiral request.
19:40.35[TK]D-FenderI am not seeing your actual full dialplan to back this up...
19:42.50CrazyTux[m][TK]D-Fender: nm, thank you - that in itself ... I found the issue ;)
19:43.34CrazyTux[m][TK]D-Fender: where do you live, let me know next time your in SoCal - I'll buy you some drinks ;)
19:44.34[TK]D-FenderCrazyTux[m], So let me guess... blacklist in the wrong context, right?
19:44.45CrazyTux[m][TK]D-Fender: ;)
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19:45.05[TK]D-FenderCrazyTux[m], Yeah, you showed 2 lines with no header... no way I'm going to trust that on-sight ;)
19:45.13CrazyTux[m][TK]D-Fender: hahahah
19:45.27[TK]D-Fender"That's nice" (tm)
19:45.33CrazyTux[m]lol
19:45.40CrazyTux[m]<-- way too hyped up right no on caffeine
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19:46.55[TK]D-FenderAnd yeah .. SoCal .... odds are in the "Not likely" category unfortunately....
19:47.30CrazyTux[m][TK]D-Fender: well let me know when you come down to Disney Land!
19:47.37Kattyi wanna go.
19:47.39CrazyTux[m][TK]D-Fender: everyone has to hitup Disney land every couple years.
19:47.39Kattyto disney land.
19:47.49Kattywhy can't i come.
19:47.57[TK]D-FenderNever been ...
19:48.11CrazyTux[m]Katty: you're more than welcome too :)
19:48.15n3hxsDisney world is better.
19:48.16[TK]D-FenderWhen I was a kid it was somewhere we were always supposed to eventually go to.... never happened
19:48.16*** join/#asterisk stupidnic (~foo@cpe-70-94-232-191.sw.res.rr.com)
19:48.22CrazyTux[m][TK]D-Fender: you should come check it out :)
19:48.29coppiceI can see Disneyland from where I am sitting. I don't feel any need to go there
19:48.33Kattywell some people just can't afford to go
19:48.36Kattyespecially now :<
19:48.44Kattyand to be honest, sea world was like a bazillion times better.
19:48.45KattyWHALE
19:48.48n3hxsTrue.
19:48.52[TK]D-FenderBut I'm already surrounded by clowns!
19:48.54stupidnicIs there an advantage of running the Cisco 7960/7940 with SCCP versus SIP?
19:48.59KattyBUT NOT BY WHALES
19:49.03[TK]D-Fenderglares at his coworkers walking by....
19:49.04Kattythat's this country.
19:49.18Kattyscowly mc scowlerson.
19:49.53ErectBananacoppice: disney land is a real pain
19:50.12ErectBananacoppice: long lines and screaming kids everywhere.  Bring your xanax
19:50.42Kattyor better yet, don't go!
19:50.46coppicetheme parks are based on getting people to queue up to the limit of what they will stand
19:50.47ErectBananathat too
19:50.55ErectBananaKatty: I'm just saying if he's "forced(tm)"
19:50.56Kattychildren aren't for me.
19:51.07Kattyno one is going to force fender to do anything
19:51.12Kattyhe'd jujitsu them to bits.
19:51.14jayteeDisney Land is the IBM XT with 640K of ram compared to Disney World being a Pentium 4 with 1GB
19:51.28Kattyso who has the i7s right now
19:51.40coppiceI can see he firework display from here every night
19:51.41Kattyand don't say curiosity
19:51.42ErectBananaKatty: I don't know him personally but I'm just saying in general the female other half will force the male half to go or be on the couch for a month
19:51.45ErectBananaetc
19:51.45Kattycause that's not what curiosity is running
19:51.59Kattywhat other female half ^_-
19:52.05ErectBananalol
19:52.06Katty[TK]D-Fender: when did you get a female half
19:52.14ErectBananaKatty: I was speaking generally
19:52.25ErectBanana;)
19:52.31[TK]D-FenderKatty, Pardon?
19:52.43Kattynevermind.
19:59.21n3hxsZooom, right over the heads. ;-)
20:00.07stupidnicIs there an advantage of running the Cisco 7960/7940 with SCCP versus SIP?
20:01.19n3hxsMaybe licensing cost?  Don't know really.
20:01.50lorsungcustupidnic: yes
20:02.03lorsungcubut afaik sip is more stable
20:03.24WIMPyIn Asterisk, not on the phone.
20:04.13WIMPy(if)
20:09.10*** join/#asterisk OliveiraBorges (~root@189-13-125-180.user.veloxzone.com.br)
20:09.15*** join/#asterisk dijib (~dijib@208-96-84-35.eastlink.ca)
20:09.16OliveiraBorgesanybody can say me what this program  -> www.viamobile.com.br/qualprograma.jpg
20:09.23jeffspeff[TK]D-Fender, thanks for that tid bit... got it figured out
20:09.32jeffspefflearned about the loca/ channel
20:09.32dijibhello all, how stable and usable is asterisk 2 now???
20:09.45stupidniclorsungcu: okay.
20:10.37dijibwait 10?
20:10.58chuckfdijib: good enough for me to use it
20:11.05drmessano10 works great
20:11.56dijibshould i upgrade?
20:11.59dijibmigrate.
20:12.24dijibim running 1.8.7 but ive discovered my fax is not working so i need to fix that and im thinking of upgrading at the same point
20:15.42navaismoOliveiraBorges, looks like http://www.asternic.biz/demo/screenshots
20:17.50drmessanoI upgraded to 10 for SIMPLE and the new Confbridge.  Was worth it
20:18.08drmessanoCan't wait until meetme is a distant memory
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20:26.45OliveiraBorgesnavaismo: is not it
20:27.09OliveiraBorgesnavaismo:  you know other sofware call center at same model
20:29.46dijibwould someone be able to explain to me beyond asterisk-10.8.0-rc1.tar.gz i should need to compile and install for res_fax / res_fax_digium? still need those?
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20:34.39jeffspeffjust so everybody knows, our phone system now does automated pages at specific times to let everybody know when break time is... the system now plays chewbacca noises. :)
20:36.08ravnxhaha thats awesome.
20:39.33anonymouz666drmessano: confbridge and t38gateway are two huge reasons to move to 10 BUT as 10 is not a LTS release... this is a huge reason for not move.
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20:43.14OliveiraBorgesanybody can say me what this program  -> www.viamobile.com.br/qualprograma.jpg
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20:43.54anonymouz666OliveiraBorges: it would be very hard to paste a link with portuguese stuff in it
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20:44.52anonymouz666OliveiraBorges: it seems a custom billing software probably based on asterisk
20:45.26CrazyTux[m]Anyone ever have a problem with asterisk not responding 200 OK to a spiraled request back to asterisk?
20:45.35CrazyTux[m](ruri changes)
20:45.36anonymouz666CrazyTux[m]: what version ?
20:45.56CrazyTux[m]anonymouz666: Asterisk 1.4.25.1
20:46.24anonymouz666did you try the latest asterisk 1.4? there's an important fix by putnopvut regarding spiralled requests
20:46.27OliveiraBorgesanonymouz666:  so you dont would know say me correct name ?
20:46.57anonymouz666OliveiraBorges: the name it's there... it's called viamobile.
20:48.16CrazyTux[m]anonymouz666: I haven't..
20:48.21CrazyTux[m]anonymouz666: let me just try it on 1.6
20:48.31anonymouz6661.6 is dead.
20:48.36anonymouz666hehe
20:49.00anonymouz6661.8 is incredible stable and great, I am using the latest 1.8.16.0-rc1 in production
20:50.00OliveiraBorgesanonymouz666: what he name ? viamobile ?
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21:17.52malcolmdnavaismo: re: DPMA & realtime.  we've thought about it.  it wouldn't work with active realtime, just static realtime, due to the way that config hooks work.  would that work for your use case?  do you just want a db back-ending DPMA, or do you need something that loads and unloads peers out of memory like active realtime?
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21:21.57Qwell11.0-beta1, whaaaaaaat?
21:22.05Qwellerr, missed a .0
21:22.29drmessanoOMG REALLY?
21:22.41mjordanRLY
21:22.44QwellMAYBE
21:22.50drmessanoEHMAGERD
21:24.56malcolmd"pull the van over honey, let's have a look at this train wreck.  kids, say cheese!"
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21:25.52drmessanoI'm excited
21:26.12drmessanoI can't wait to kick the tires on this new stuff that will make the other stuff we used for google voice look like crap stuff
21:26.37drmessanochan_not_jingle?
21:26.41Qwell"look like"...  right...
21:28.02stefanskiI always get ./configure error with several versions of asterisk : ./configure: line 25487: 2*10000+10*100+6 ((major<<24)|(minor<<16)|build) >=: syntax error in expression
21:28.10parasitodelsurthis stuff is awesome
21:28.28malcolmdchan_motif seemed to work :D   couple of functional changes though for folks...indications aren't automatically generated, so you'll have to do that yourself when you Dial(), and you'll want to Wait a sec before answering so that so that google doesn't think the call is still being setup.
21:29.13drmessanoNoted
21:29.51malcolmdthe calling using google wiki page has been updated to reflect those changes, so give it a quick once-over and you'll be just fine
21:30.43malcolmdalso, username for the xmpp definition no longer has the /Talk suffix
21:30.54stefanskiany ideas what's causing the syntax error?
21:31.18drmessanoI never used the /Talk resource.  Wasn't needed
21:31.18malcolmdand be careful with codecs since google talk supports g.722 and google voice doesn't..google voice will happily setup a g.722 call, you just won't have any audio
21:31.32drmessanoNice!
21:31.58malcolmdo yea, it was a "what tha" moment
21:32.38drmessanoGV supports g.711, g.723, and gsm, or something like that.. IIRC
21:32.44Qwellstefanski: What version[s] of Asterisk?
21:32.56navaismomalcolmd, yeah i only want db  back rnding
21:33.10navaismostatic its ok fo me
21:33.36malcolmdgoogle says one thing is supported, but doesn't actually add capabilities for what it says it should.  we see PCMA, PCMU, G.722, iLBC, Speex @ 8 and 16kHz
21:33.51malcolmdthey're supposed to do GSM, but i haven't seen it
21:34.04malcolmdnavaismo: cool, thanks for the feedback :)
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21:34.25stefanski1.8.15
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21:34.32navaismothan you
21:34.49stefanskialso with 10.7.0
21:35.27drmessanoInteresting
21:36.19navaismoby the way malcolmd  great work on DPMA m customer  love it really
21:36.45Qwellmalcolmd is the best product manager evar.
21:36.47malcolmdprops to Qwell for all development.  i'm just the idiot product manager :)  if something's hard to do, it's my fault, not his. :D
21:36.50QwellDon't tell him I said that.
21:37.21malcolmdQwell: you're just trying to butter me up for more release beers when we finish 1.3 ;)
21:37.27QwellYou are correct.
21:37.45anonymouz666beer time, bye
21:38.08malcolmdnavaismo: thank you for your help on the forums.  we both had a hard time with the cuban_cigar person.  i haven't heard back from them; i hope they succeeded
21:38.31mjordanstefanski: if you can't figure out what's up, please file an issue.  Make sure you include your environment, as its highly likely to be related to that
21:38.50mjordana few of us are about to head out for the evening, but I don't want a configure issue to drop
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22:00.38parasitodelsuryay kerner trace
22:00.40parasitodelsurso cool
22:00.45parasitodelsurnumbers everywhere
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