IRC log for #asterisk on 20130722

00:06.37*** join/#asterisk Ionic (~ionic@2001:41d0:1:6276:fa1e:dfff:fedd:1f7)
00:07.39Ionichow does asterisk actually select the codec to be used for a call?
00:07.58Ionicis it trying to determine the best codec of all involved peers/channels?
00:20.33pabelangerIonic, it is a negotiation process between asterisk and the end point, each will have a list of codecs and order.  Match is made
00:21.58Ionicpabelanger: yes, kinda, I guess
00:22.41Ionicpabelanger: only in my case, I'm going through a middle point (ISDN), which is only capable of ulaw 8KHz PCM
00:22.57pabelangerthen that is all you will use
00:22.58Ionicso I guess I'm stuck with that
00:23.24Ionicyeah, makes sense
00:24.54IonicI thought maybe asterisk was capable of receiving speex32 from my SIP client and converting it back and forth, but that doesn't make a lot of sense, better use ulaw straight through
00:26.23WIMPyYou could use G.722 if you still have a real phone line and someone had implemented it in any of the Asterisk channels.
00:29.57Ionic100012 audio     g722 (G722)
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00:31.38volga629is possible hash out body  message like Fee  Set("Message/ast_msg_queue", "BASE64MSG=BASE64_ENCODE(Fee)") in new stack
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00:37.01Ionichmm, but if I knew whether that's supported by the channel
00:39.23Ionicshould be, though
00:41.01florenis format_mp3 still dependent on mpg123 for asterisk 11? i thought is not, i geta compile error
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00:46.26Ionic(gsm|ulaw|alaw|speex|speex16|g722|h264|speex32) hmmm... not exactly the right order
00:46.27ChannelZyes
00:46.34Ionicand stuff like opus is missing too
00:46.44ChannelZWho uses it?
00:47.13IonicI defined it as the top/preferred codec in sip.conf/users.conf
00:47.27Ionicso, even while not using it, I'd expect asterisk to put it in front of the list
00:47.28ImpossStrange issue: I can call  setup my voicemail box, leave messages in my voicemail box, but when I hit the voicemail button enter my password "You have no messages"
00:48.06ImpossSystem is Asterisk 11, FreePBX latest version
00:48.09ChannelZhave they even finished opus for asterisk?
00:48.56Ionicnevermind me, core show codecs does not list opus
00:49.01ChannelZthere's silk, if you go download it, but not opus AFAIK
00:49.04Ionicstill... the list is pretty much in the wrong order
00:51.19Ionicallow=opus,speex32,speex16,speex,g722,ulaw,alaw
00:51.20Ionichmhm
00:52.30Ionicwell, I could try disallow=all first
00:53.44Ionicmeh, now GSM is disabled, but the order is still messed up
00:53.59Ionicchan_sip.c:13084 add_sdp: ** Our capability: (ulaw|alaw|speex|speex16|g722|h264|speex32)
00:57.58Ionicoh, great
00:58.07Ionicdisallow = all in users.conf is giving me speex32
00:58.12Ionicwait.. what?
00:58.34volga629yes, got fixed exten => _X.,n,Set(BASE64MSG=${BASE64_ENCODE(${MESSAGE(body)})})
00:58.57volga629thank you Everyone for hints and help
00:59.29IonicOur capability: (ulaw|speex|speex16|g722|speex32)
00:59.49Ionicand then speex32 gets selected... that's not making any sense, UNLESS the last entry is the most significant one
01:00.14Ionicbut then again, why was ulaw selected before
01:01.18Ionicchan_sip.c:7911 sip_new: *** Our native formats are (speex32) < hum...
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01:03.07Ionicprobably because my client prefers speex32... but... oh well
01:03.07Ionic*moving g722 up*
01:07.14Ionicwell, that didn't help either...
01:07.26IonicI'm confused, but at least it's using speex32 now
01:15.06WIMPyIIRC it makes a difference if you use multiple allow lines instead of one with a list.
01:16.36IonicI'm now using multiple disallow lines as well
01:17.03IonicConfused, but stopping right here, I've found some worse problem than encodings
01:17.16WIMPyThe usual story.
01:19.49Ioniccalls keep hanging around in the capi channel list even after they are terminated/disconnected and a hangup request won't work either, gotta sort this out first
01:20.11Ionicreloading the channel helps, but that's merely a workaround
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02:30.37floreni have a quick question related to compiling asterisk with a custom /var/lib location
02:31.43florenmake ASTVARLIBDIR=/usr/share/asterisk will not work for example
02:32.21florenwhat is the proper variable to use? I'm looking at the source code and I see a lit of references in this format: AST_VAR_DIR
02:33.08Ionicwhy would it?
02:33.18Ionicuse the correct configure switch, --libdir
02:33.25Ioniccheck ./configure --help
02:33.40florenya
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02:55.24florenis not working, i set %configure --sharedstatedir=/usr/share instead of /var/lib and it still compiles into /var/lib
03:19.58Ionicwth
03:20.09Ionic--sharedstatedir isn't even used by asterisk I guess
03:20.33Ionicwhy don't you just set --libdir=/usr/share?
03:20.44Ionicand why would you have libraries in /usr/share anyway?
03:20.59Ionicthat's not sane and against any established FSH
03:22.43Ionicsharedstatedir is not used by *any* makefile as far as I can see
03:23.53Ionicplease, I beg you, don't try to change options you don't understand, this will only lead you to broken system...
03:24.31Ionics/lead you to/lead to a/
03:29.22WIMPyhuh?
03:30.00WIMPyThe great things about unixoid OSs is that you can change almost everything. What's your issue with that?
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03:47.42IonicWIMPy: in theory, yes, but first and foremost /usr is vendor land and then there are file system hierarchy rules to make life less of a pain
03:51.59Ionicanyway, I'm stepping to bed, night and good luck
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05:53.06hrolfHi #asterisk, how can we do a supervised (2 step) transfer in Asterisk?
05:54.47WIMPyCheck your terminals manual or features.conf.
05:55.05hrolfWIMPy: What is terminals manual?
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05:56.06WIMPyTerminal= Phone or ATA ore whatever you might be using.
05:56.34hrolfWIMPy: I'm using SIP
05:56.48hrolfWIMPy: SIP trunk between Avaya and Asterisk
05:57.07WIMPyThen check the Avayas manual.
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06:04.20hrolfWIMPy: But I want to do it from Asterisk's side not Avaya.
06:04.30WIMPyWith what?
06:05.45hrolfWIMPy: With the SIP trunk established with Avaya. That is, call is received on the SIP trunk from Avaya on Asterisk. Now do a 2-step xfer from Asterisk to an extension in Avaya.
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06:06.52WIMPyI'd use a phone.
06:07.33WIMPyCheck your terminals manual or features.conf. <= two possibilites to do transfers on Asterisk.
06:08.38hrolfWIMPy: One question, how does Asterisk do the transfer internally? Does it uses Transfer()?
06:08.56hrolf.. in case of 2-step xfer.
06:09.28WIMPyIf you want to send a call to another URI, you can use Transfer().
06:10.16WIMPyIf someone wants to do it manually, he will use his phone or DTMF features. That's not dialplan.
06:12.21hrolfWIMPy: I want to do it through the dialplan, so in that case I believe features.conf doesn't have anything to do, right?
06:13.17WIMPyCorrect, but doesn't fit your original question.
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06:14.31hrolfWIMPy: Means that 2-step transfer is only for phones, not dialplan?
06:14.41WIMPyTransferring autonatically and attended don;t go together.
06:15.08WIMPyHow are you going to do that if you only have a caller and no callee?
06:17.19hrolfWIMPy: I was under the impression that, we can do 2step transfer in dialplan like, Dial(), then after the called party answers, use function XXXXX() to complete the 2-step xfer?
06:17.57WIMPyThe dialplan won't continue before one party hangs up.
06:18.10hrolfWIMPy: I see.
06:18.52WIMPyAnd even that will only happen if you pass the right parameter to Dial().
06:20.02hrolfWIMPy: the 'g' paramtere?
06:21.10WIMPyyes
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06:21.51hrolfWIMPy: Okay so what will be that XXXXX() application/function which will then complete the 2-step xfer?
06:22.05WIMPyYour phone.
06:22.27hrolfWIMPy: There is no phone, we are in dialplan.
06:22.28WIMPyIt starts as a 2nd call.
06:22.50WIMPyThen you're not talking to someone.
06:23.07WIMPyWhat do you want to do?
06:24.23hrolfWIMPy: Call received from Avaya on Asterisk (through the SIP trunk), play IVR, then do a 2-step xfer to an extension on Avaya.
06:24.44hrolfWIMPy: We tried Transfer() (blind transfer) but it is not working.
06:24.58WIMPyWhere is the 2nd step when there is only one person involved in the call?
06:26.06WIMPyTransfer() tries to redirect the user away from Asterisk to anothr URI. If you just want to send the caller to some other device, use Dial().
06:27.01hrolfWIMPy: Please some clarification, you mean to say 2-step transfer only happens when 2 persons are involved?
06:27.29WIMPyDoesn;t make any sense to me otherwise.
06:27.52WIMPyBut maybe you should explain what "2-step transfer" means to you.
06:30.14hrolfWIMPy: (Sorry if I'm using the wrong terminology), but what I mean by 2-step xfer, is that a call is received on Asterisk, we do Dial(), the dialed party picks up, then the call is handed over to the dialed party, the one who intiaited the dial drops.
06:30.46WIMPyBut there's no call left then.
06:31.16hrolfWIMPy: There is, the original caller.
06:31.36WIMPyAnd who else?
06:31.57hrolfWIMPy: And the one to whom we just 2-step xfered.
06:32.37WIMPyMaybe you should start all over again. That doesn't make any sense at all.
06:32.49WIMPyYou said the caller is in an IVR.
06:33.29hrolfCall received from Avaya to Asterisk on SIP trunk. We played the IVR, now IVR wants to transfer to an extension on Avaya. So we can do Transfer(), or Dial().
06:33.33hrolfTransfer() is not working.
06:33.48hrolfDial() uses another channel (we don't much licenses.)
06:34.13hrolfSo I'm saying we can try the other method 2-step transfer to give the call back to an extension on Avaya.
06:35.01WIMPyThen Transfer() is your only option.
06:35.19WIMPyBut maybe the Avaya won't accept that.
06:35.26hrolfWIMPy: Yes.
06:35.36hrolfWIMPy: Do you know why?
06:36.05WIMPySecurity configuration?
06:36.11WIMPyJust not implemented?
06:36.46WIMPyMaybe it's cheaper if you connect them via PRI instead of SIP?
06:37.32hrolfWIMPy: Okay.
06:37.41hrolfWIMPy: The customer will have to buy a card then.
06:38.07hrolfWIMPy: One thing, can you explain what is a 2-step xfer. Is it even possible in the scenario I described?
06:38.43WIMPyI just set up an Asterisk with PRI card for someone who wanted to use SIP with his PBX. Two PRI cards cost less thatn the SIP licenses.
06:39.08WIMPyI still don;t know what you mean by 2-step transfer.
06:40.44hrolfWIMPy: Like from in the dialplan, after playing IVR, you do another Dial(,'g'), the dialed party answers, the caller (who called the IVR) is transfered to that dialed party.
06:41.49WIMPyUsually the underlying channel will do so automatically if possible and configured.
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06:42.34WIMPyBut I'm not sure if that would be media only for SIP. I never dared to try.
06:44.47hrolfWIMPy: Okay. Can you tell what is the propery format for Transfer()?
06:45.37WIMPyA full URI.
06:46.05hrolfWIMPy: Is it Transfer(sip/avaya/252) where sip/avaya is the SIP user I have defined in my Asterisk (with it's host=<IP Addr of Avaya) and 252 is the extension on Avaya?
06:46.15hrolfWIMPy: On version 1.6
06:46.37WIMPyBut if I read the info correctly, that would only do what you want if the call hasn't been answered yet.
06:47.18WIMPyAFAIK it's a URI, not a peer.
06:47.52WIMPysip/destination@avaya.local or something, but see above.
06:48.07hrolfWIMPy: The documentation for Asterisk 11, says Transfer(Tech/destination), isn't that Transfer(SIP/avaya/252) ?
06:48.53WIMPyI'm pretty sure I read that that does not work.
06:49.20hrolfWIMPy: I have tried Transfer(sip/avaya/252) between two Asterisk servers (assuming avaya's host=<IP Addr of the other Asterisk>) and it worked.
06:49.30hrolfWIMPy: But a full URI didn't
06:49.35WIMPyBut maybe it will work if the call was answered, but then it will use two channels as I understand that.
06:49.48hrolfWIMPy: and no two channels were used.
06:50.16WIMPyWell, ok, then you know how to do it.
06:50.52hrolfWIMPy: Yes, but I thought may be due to the wrong format it is not working with Avaya etc.
06:51.18WIMPyYou can always go and read the sip debug.
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07:02.00hrolfWIMPy: Can you take a look at the logs? I have both from avaya's side and Asterisk's side. (SIP debug)
07:02.07hrolfhttp://pastebin.com/Z62QMm5G
07:02.11hrolfhttp://pastebin.com/27PksdbT
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08:38.45tparcina"voicemail show users for default" shows that users have messages, but there are no dirs in /var/spool/asterisk/voicemail/default/
08:38.50tparcinaHow can this be?
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08:42.55SchredaHEy everybody
08:43.17SchredaI would need some help and it would be nice to get an feedback :D
08:44.17SchredaIs it possible to have 2 different dialplans but with same extensions?
08:44.18Schredae.g
08:44.56SchredaI have  one phone which can be called by using an extension e.g 1
08:45.23Schredaafter some time and nobody answered this call is forwarded to other phones ... This works fine
08:47.28SchredaThe problem now is that I get a call on a other phone and if this phone isnt answered I call all phones incl. extension 1... but now an dial circle exist because if the extension 1 is not reply after a certain time... So I want to use 2 different extension configuration depending on which extensions got called...
08:47.32Schredais it possible?
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10:33.13MadsHello, i'm having a problem with the Asterisk cdr csv file. As far as i can see the csv file contains 16 fields my problem is within field 3 (destination). When call forwarding is enabled the destination is not showing the number that is forwarded to but the number that is called in first place, is this the normal way it works or some bug?
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11:16.28LooserOutingdoes asterisk support ilbc mode 20
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12:00.00kchehabhi i am using 1.8.16 , trying to make the rtp of calls pass by my asterisk i set in sip.conf directmedia=yes directmediadeny=0.0.0.0/0 canreinvite = no
12:00.07kchehabbut still not working please  advice
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12:15.23kchehabplease any one can help ?
12:16.50Guggekchehab: if you dont want directmedia, why do you set it to yes?
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12:33.00MadsHello, i'm having a problem with the Asterisk cdr csv file. As far as i can see the csv file contains 16 fields my problem is within field 3 (destination). When call forwarding is enabled the destination is not showing the number that is forwarded to but the number that is called in first place, is this the normal way it should be?
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12:44.47Greenlightkchehab: You want RTP to bypass your Asterisk box? For this to happen the codecs etc need to match exactly
12:46.25GreenlightMads: Yes
12:49.42kchehabGameGamer43 yes i want that
12:50.07kchehabGugge soeey i set it to no in my config ,this was a mistake i fix it and still have the problem
12:50.29kchehabGugge RTp is still oassing directly ,and i am debugging using rtp set debug on
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12:50.44GreenlightSo there are a number of things that will prevent direct RTP
12:51.00GreenlightAre you call recording? Are you detecting DTMF? Are the codecs different?
12:51.48kchehabGreenlight no
12:52.16kchehabGreenlight is my config has a right syntax
12:52.40Greenlightdirectmedia=yes on the peer(s) is all that's needed
12:53.00jmetroor directmedia=no. some people prefer that.
12:53.04kchehabGreenlight   using sip show setting i can see   Direct RTP setup:       No
12:53.26Greenlightjmetro: If the goal is to enable direct media, I don't see why you'd set that ....
12:53.41kchehabGreenlight i will set directmedia=no on user and trunk
12:53.44kchehaband try
12:53.51GreenlightUmm
12:53.57GreenlightSure... good luck with that.
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12:54.56jmetroif you WANT it, then directmedia=yes indeed
12:55.03jmetroit seems like it causes issues though
12:55.12kchehabGreenlight i think it works yes
12:55.18kchehabthanks alot
12:55.28Greenlightshurgs
12:57.42MadsMy problem is that the guy who is forwarding his phone needs to be billed for the forwarding call but the CSV file is just logging the call as a standard incomming call that should not be billed.
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13:00.03GreenlightMads: Are you sure you don't get a *second* CDR for the forwareded call ?
13:00.14ThothCastelwhy do I keep on getting 'failed for 192.168.100.46:33992' - wrong pasword' error message when trying to register a sip account user (Zoiper Communication' softphone
13:00.46eirirs_network problems
13:00.47GreenlightThothCastel: Going to take a wild stab in the dark guess here - are you using the wrong password ?
13:00.55eirirs_haha
13:00.59MadsGreenlight there is no second CDR for the forwarded call
13:01.00ThothCastelGreenlight: NO
13:01.13ThothCastelsorry, meant no
13:02.01GreenlightDoes the account exists and you've reloaded if neccissary ?
13:02.02ThothCastelGreenlight: I added [1001]
13:02.09ThothCastelusername=1001
13:02.18ThothCastelsecret=QwEr1QwEr2q
13:02.31ThothCastelall these to the end of the sip.conf file
13:02.48GreenlightYou're *sure* you've reloaded ?
13:02.52ThothCastelreloaded? well, I reboot the OS
13:02.58GreenlightOkay
13:03.17GreenlightAnd you've not mistyped the password ?
13:03.59*** join/#asterisk italorossi (~italoross@187.60.66.11)
13:04.56GreenlightMads: I would have expected a 2nd CDR. Perhaps if you enable unanswered calls CDR's it will show
13:05.10GreenlightUnless - how are you doing the redirect ?
13:08.38ThothCastelGreenlight: do I need to add the 1001 extension to the extensions.conf file as well?
13:09.16GreenlightNot for it to register, no.
13:09.35GreenlightLets see your sip.conf
13:09.38Greenlight~pb
13:09.38infobotA "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood 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 install pastebinit with yum or aptitude.
13:09.41Greenlight^^^
13:10.42MadsGreenlight the call forwarding is done through an extension, if a forwarded call is unanswered it does show the unanswered call as expected :)
13:11.20ThothCastelGreenlight: http://pastebin.com/8XaUEXGN
13:12.26Greenlighttupe=friend
13:12.30GreenlightThat should be type
13:12.35GreenlightFor starters
13:12.41jmetros/tupe/type
13:13.45*** join/#asterisk slav3_kitten (~frankthet@unaffiliated/slav3-kitten/x-0866809)
13:14.11GreenlightFix that, and I bet it'll work
13:14.55*** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan)
13:15.25ThothCastelGreenlight: thank you, I updated and did a 'core restart now' and then 'asterisk -r'
13:15.41ThothCastelI tried registering the softphone now and still the same error msg
13:16.32*** join/#asterisk Cuzner (~ccuzner@198.41.29.45)
13:17.35ThothCastelGreenlight: this is the error msg http://pastebin.com/iAbHyesK
13:17.44GreenlightOh. Add type=friend and context=LocalSets, host=dynamic
13:17.58GreenlightDIdn't spot that it wasn't inheriting
13:19.47ThothCastelGreenlight: these are already added lines 11, 13 and 15 respectivelly
13:20.06GreenlightYou're trying to register 1001 though
13:20.49ThothCastelright, I see so I should add those to each of the [device] ?
13:21.05GreenlightOr add (office-phone) after the [1001]
13:21.12GreenlightTo get it to inherit the template
13:21.14ThothCastelwon't [1001] get the settings from the (!) template?
13:21.51GreenlightIf you tell it to, yes
13:22.10GreenlightAs you have with your other devices
13:22.21ThothCastelyey! Thanks Greenlight. it worked now that I added the (office-pone) to it
13:22.29ThothCastelIt registered no
13:22.32ThothCastelnow
13:22.35GreenlightExcellent
13:22.52*** join/#asterisk sp00kz (~ilubj00@unaffiliated/sp00kz)
13:22.57*** join/#asterisk jeev (~j@unaffiliated/jeev)
13:23.13ThothCastelGreenlight, so on the way it is now, if I register a second softphone with a different extension number, then I will be able to place a call from ext 1001 to ext 1002 (eg)
13:23.15ThothCastel?
13:23.56ThothCastelOr I still need to configure some other stuff?  this is my first time setting up asterisk
13:24.22GreenlightYou would need to configure that in your extensions.conf
13:24.46GreenlightSpecifically in the LocalSets context that you've set them to go to
13:25.24GreenlightAt a very basic level, you could define exten => _XXXX,1,Dial(SIP/${EXTEN})
13:25.31Greenlightsame => n,Hangup
13:25.40*** join/#asterisk Rumbles (~Rumbles@77.107.183.230)
13:25.51GreenlightSo when a 4 digit number was dialled it'll try and find a matching SIP peer
13:27.21*** join/#asterisk Rumbles (~Rumbles@77.107.183.230)
13:27.40ThothCastelGreenlight: right, where EXTEN = 1001?      or where XXXX = 1001
13:28.56*** join/#asterisk mjordan (~mjordan@nat/digium/x-nynxbbzttjobnfeh)
13:28.56*** mode/#asterisk [+o mjordan] by ChanServ
13:29.12GreenlightThe _XXXX will act as a "catch" for any 4-digit number dialled. ${EXTEN} will be subsituted with that 4 digit number.
13:32.12*** join/#asterisk bulkorok (~chatzilla@85.183.61.47)
13:33.10bulkorokhi... how can I configure Asterisk that SendFax() send RTP after receiving 183 SDP and not only after 200 OK ?!
13:34.41jmetrohttp://lmgtfy.com/?q=asterisk+send+rtp+after+receiving+183&l=1
13:35.21jmetroi dont know anything about fax.
13:35.28jmetrobut i found your answer.
13:36.01ThothCastelGreenlight: thank you :)
13:41.56*** join/#asterisk vlad_starkov (~vlad_star@195.239.220.131)
13:43.32vlad_starkovQuestion: Anyone know is it possible to retrieve callerid information from parked call in exact parking lot?
13:44.01jmetrovlad_starkov: yes.
13:44.11vlad_starkovjmetro: could you point me please
13:44.12*** join/#asterisk leedm777 (~leedm777@nat/digium/x-yjsvvlhfrvfgirrd)
13:44.43jmetrovlad_starkov: i know Aastras can do it, via XML apps.
13:44.45vlad_starkovjmetro: what I need is to retrieve callerid of parked call before picking up that call
13:45.11vlad_starkovjmetro: I mean to do it in * dialplan
13:45.13jmetrovlad_starkov: i actually have a button on my phone that shows the first 9 parked spots, caller ID's
13:46.50*** join/#asterisk Rumbles (~Rumbles@host-92-27-117-96.static.as13285.net)
13:49.05jmetroI dont know how to do it in dialplan, but on the phone is kind amore useful isnt it?
13:52.48vlad_starkovjmetro: it is useful to make it in dialplan when you write some logic
13:53.32bulkorokjmetro: the r is not set in my Dial
13:55.27*** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
13:55.28*** mode/#asterisk [+o putnopvut] by ChanServ
13:55.55*** join/#asterisk btracht (~btracht@70.89.37.217)
13:56.34*** join/#asterisk Caplain (~shayne@2604:8800:123:0:a8dc:ede4:af2a:eca6)
14:03.11bulkorokso... anither idea why sendfax does not begin rtp after 183 but after 200 ?!
14:04.59*** join/#asterisk Katty (~Katty@mail.copi-rite.com)
14:05.11Kattyhello my asterisk does not work at all how to fix plz is urgent thx
14:05.40jmetroKatty: go to www.
14:05.51Kattyjmetro: what is www. plz
14:06.00jmetrowww.girlsgogames.com/‎ they have manuals for fix urgent plz
14:06.07btrachti am running trixbox v 2.8.0.4 and trying to decipher the CDR report. what does it mean when the channel is SIP/the servers own ip?
14:07.14*** join/#asterisk Korrosion (~nate@70.89.37.217)
14:08.03Kattyhugs on jmetro
14:08.08jmetro:3
14:08.14jmetroMornin lass
14:08.24Kattyhow's things out your way?
14:08.32btrachtor when the source is root?
14:08.32*** join/#asterisk hexanol (~bibi@modemcable094.94-70-69.static.videotron.ca)
14:08.41jmetrodead hot. But air conditioned.
14:08.48Kattyyay air conditioning!
14:09.21*** join/#asterisk Qwell (~north@pdpc/sponsor/digium/Qwell)
14:09.21*** mode/#asterisk [+o Qwell] by ChanServ
14:09.31igcewielingbtracht: Could be a number of reasons.   Unfortunately, not many people will want to deal with Trixbox
14:09.35*** join/#asterisk jsjc (~Adium@178.Red-79-150-247.dynamicIP.rima-tde.net)
14:09.57btrachtnumber of reasons such as?
14:10.25btrachtwhy not deal with trixbox?
14:10.31Kattythere is a channel for trixbox
14:10.39Kattyit's different.
14:10.44igcewielingbtracht: someone hacked into your server, you don't have allowguest=no or you have a config issue on your peers or the call had some kind of off forwarding or routing
14:10.57Kattyi blame squirrels.
14:10.58igcewieling~trixbox
14:10.59infobotDelving into Trixbox is like exploring a pyramid; it's ancient, forgotten, dark, and dangerous.  Trixbox was one of the earliest complete PBX distros and a relic of a bygone era.  While it was a great idea, it was implemented by a horrible group of Wizards from an evil, barren wasteland that stuffed it full of black magic and FUD.  Also, an example of how not to run a business.
14:12.28GreenlightMy favourite infobot description ^^
14:12.56btrachtwe are in the process of converting from trixbox but in the mean time i am trying to figure out whats going on from the current server
14:14.23[TK]D-Fendertrixbox died about 3 years ago
14:14.32[TK]D-FenderAnd they ran forked versions of Asterisk and FreePBX.
14:14.35[TK]D-FenderNone of which we support
14:14.42slav3_kittenmay it rest in peace
14:18.35jmetrobtracht: look at the UI and make a flowchart.
14:18.40jmetrobtracht: then hand-code it.
14:18.51jmetrobtracht: or just call yourself and see what happens, flowchart, code.
14:19.25[TK]D-Fender[10:10]igcewielingbtracht: someone hacked into your server, you don't have allowguest=no or you have a config issue on your peers or the call had some kind of off forwarding or routing
14:19.46[TK]D-FenderThis was prety much the answer though as it was a CDR question.  That is simply what the channel looked like when it came in
14:20.08[TK]D-FenderQuite possibly a bad trunk setup failing over to an un-authed call
14:25.45*** join/#asterisk serafie (~erin@206.255.84.113)
14:25.59igcewielingbtracht: you'll need to comb through the Asterisk logs to find the problem call and see what actually happened to cause the issue.
14:27.39Kattygood morning fender bender. did you have a good weekend?
14:28.15bulkorokjmetro: another idea for no RTP after 183 ?!
14:28.19*** join/#asterisk vlad_starkov (~vlad_star@nat.canmos.ru)
14:28.48jmetrobulkorok: probably google, i dont know anything about fax at all.
14:29.18bulkorokjmetro: k... google is not that helpfull in that case
14:29.38igcewielingfax does not use RTP, so your question doesn't make a lot of sense.
14:30.10bulkorokwell how are the "fax tones" aka CNG transmitted!?
14:31.26*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
14:31.26igcewielingbulkorok: using RTP, but that is not actually fax.   Have you confirmed other applications do not exibit the same issue?  Have you tried the z option to SendFax?
14:31.54*** join/#asterisk italorossi (~italoross@187.60.66.11)
14:35.11igcewielingno RTP after a 183 sounds like a more general sip.conf issue rather than something specific to sendfax.
14:36.41*** join/#asterisk jasonwert (~w3rt@75-134-81-98.static.aldl.mi.charter.com)
14:36.49bulkorokI didn't try z opetion...
14:37.15bulkorokDial has not the r option (not any option...)
14:37.46CuznerThat trixbox bot description is great
14:38.28CuznerI used to set-up trixbox's for small/medium businesses, it did what it was supposed to do...
14:38.46Kattypokes [TK]D-Fender
14:38.46igcewielingThe "r" option to dial is not useful in any case where you are not getting ringing when you should.
14:38.57bulkorokagree
14:39.11igcewielingCuzner: We've all done things in our lives which we are not proud of.
14:39.24Cuznerigcewieling: I never said i was ashamed
14:39.26bulkorokbut which option changes the 183 behaviour?!
14:39.39Cuznerit was good for what it was, quickly set-up PBX in a box
14:39.40igcewielingCuzner: You didn't have to.  We are ashamed enough for you.
14:39.46[TK]D-Fenderprods Katty
14:39.58Cuznerasterisk 1.4 + FreePBX + whatever the hud was called
14:40.14Katty[TK]D-Fender: how was your weekend, dear?
14:40.15Cuznerand it did what it was supposed to do, i never had a problem with it.
14:40.26Cuznerit was better than everything else at the time
14:40.33CuznerElastix? wtf was that...
14:40.49igcewielingbulkorok: nothing obvious in sip.conf.sample included in your Asterisk source code?  like something to do with progress and early media?
14:40.50leifmadsenCuzner: it's what I wrapped your balls in
14:40.58leifmadsenoh shit, that wasn't a privmsg?!
14:40.58Cuzneroh my
14:41.07leifmadsencarry on! nothing to see here!
14:41.26Cuznerleifmadsen: can i get your signature? :)
14:41.31leifmadsensure! :)
14:41.31igcewielingadds another checkbox next to "leifmadsen"
14:41.33leifmadsen$10
14:41.42leifmadsenigcewieling: gold star?
14:41.52KattyLEIF
14:41.56leifmadsenKATTY
14:42.00*** join/#asterisk zrzerenato (bd2962c6@gateway/web/freenode/ip.189.41.98.198)
14:42.01Kattyhello, dear
14:42.01leifmadsenKatty: fudge. make it.
14:42.07zrzerenatoim looking forward to change a priority on certain port that i use, example: set priority highest port (8765) TCP. ????
14:42.11Kattyleifmadsen: why?
14:42.12leifmadsenKatty: deliciousness ensues.
14:42.16Kattyleifmadsen: it's easy you know. you could make it
14:42.21leifmadsenKatty: indeed.
14:42.25leifmadsenI'm just saying, you won't regret it
14:42.26Kattyleifmadsen: oh, so you've had it? excellent.
14:42.29CuznerLeaf pronounces his name Life, btw. You all have been saying it wrong this whole time.
14:42.33igcewielingzrzerenato: Try /join #TheDistroYouUse
14:42.35Kattyleifmadsen: oh, no. i wouldn't. well. kind of. lots of calories :<
14:42.42Kattyleifmadsen: but i have made it before! thank you for thinking of me, regardless
14:42.45leifmadsenKatty: ikr? :)
14:42.54Kattyleifmadsen: if you run across anything else tasty, let me know.
14:43.04leifmadsenKatty: ok!
14:43.11leifmadsenKatty: rum cookies
14:43.14Kattyleifmadsen: that miniature hooman of yours is growing up so fast!
14:43.20Kattyleifmadsen: link?
14:43.20jmetroblueberry baked goods. i dont care what.
14:43.28leifmadsenvanilla rum balls :D
14:43.30jmetroblueberry pound cake, muffins, scones.
14:43.31Kattyjmetro: i've got a killer blueberry muffin recipe, if you're interested.
14:43.39leifmadsenKatty: I just used google
14:43.42leifmadsenKatty: he is huge now....
14:43.44Kattyleifmadsen: ah, right. ok
14:43.45btrachtigcewieling: this is one of the problem calls in our asterisk log - http://pastebin.com/UshLqmMQ
14:43.58Kattyleifmadsen: ohoh
14:44.05Kattyleifmadsen: i found HobNob cookies!
14:44.13Kattyleifmadsen: do they have those up there in Canada?
14:44.18zrzerenatoigcewieling: srry , but im trying there already but no answers. And it is asswell about connections between cti, and starphone,. im having lots of connection_lost, with my asterisk.
14:44.19leifmadsenI DON'T KNOW BUT MAYBE
14:44.27[TK]D-Fenderbtracht: Your r call is not matching a trunk you set up.
14:44.32Cuznerwe have poutine up here!
14:44.33[TK]D-Fenderbtracht: So fix your trunk
14:44.35Cuznerthat's all you need
14:44.38Kattylocates a photo
14:44.39igcewielingbtracht: looks pretty obvious to me.  Received incoming SIP connection from unknown peer to +972598461260") in new stac
14:44.46*** join/#asterisk TimeRider (~steve@host81-136-216-215.in-addr.btopenworld.com)
14:45.12Kattyleifmadsen: http://ukonlinegoodies.com/images/mcvitties-dark-choc-hobnobs-300g.jpg <- addictive.
14:45.25igcewielingzrzerenato: English is not your native language?
14:45.26bulkorokigcewieling: I used the sample sip.conf... allowoverlap=no directmedia=yes ignoresdpversion=yes
14:45.37btrachtin our CDR the channel is SIP/servers ip address, source is root and CLID is "root"<root>
14:45.52igcewielingbtracht: correct.  Because there is no peer for that call
14:45.59jmetrobtracht: whats the destination? what is it doing?
14:46.28zrzerenatoigcewieling: nope, but it is readable? rsrs
14:46.39igcewielingbtracht: try prematuremedia and progressinband options.
14:46.42[TK]D-Fenderbtracht: Fix your trunk
14:46.56igcewielingzrzerenato: we have no idea what you are talking about.
14:46.56[TK]D-Fenderigcewieling: It isn't matching...
14:47.16btrachtwhat do you mean by fix your trunk? other calls work. could it be because this is an international call?
14:47.21igcewielingsorry ths is for bulkorok: try prematuremedia and progressinband options.
14:47.32[TK]D-Fenderzrzerenato: You seem to be asking a core networking question.  This is something you should ask in your distro's channel
14:47.38bulkorokigcewieling: btracht: try prematuremedia and progressinband options. <= is that for me?!
14:48.24igcewielingbtracht: add "allowguest=no" and "alwaysauthreject=yes" to your setup.  that should fix the issue.
14:48.26Cuznerleifmadsen: your comment made me giggle, but i'm in a video conference with Tom right now, so not the best timing :P
14:48.39leifmadsenCuzner: I disagree
14:48.58Cuzneryou'd argue it was perfectly timed, i know.
14:48.59zrzerenato[TK]D-Fender: thats correct. so it is understanding . my distro is centos, and im trying to ask there but seems no one is online, coz there is not chat on the channel
14:49.02btrachtigcewieling: to my trunk details?
14:49.02[TK]D-Fenderigcewieling: Nope, won't make his peer match any better... he needs to fix that direct
14:49.16[TK]D-Fenderbtracht: Make your trunk match
14:49.27igcewielingbtracht: no.  to sip_general_custom.conf
14:49.38[TK]D-Fenderbtracht: verify the auth your provider is sending (or not) with their calls to you ... and set your trunk accordingly
14:49.45igcewieling[TK]D-Fender: he doesn't understand enough to make is "trunk match",
14:50.04igcewielingonce he adds the options I recommended and some of his calls break, THEN he will be ready to fix his trunk.
14:50.34[TK]D-Fenders/some/all
14:51.52*** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
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14:52.04KorrosionWhat is he matching the trunk to?
14:57.04[TK]D-Fenderit ISN'T matching.  That is the problem
14:58.18Korrosionmatching what though?
14:59.00[TK]D-FenderMatching the CALL clearly...
14:59.59*** join/#asterisk Rumbles (~Rumbles@77.107.183.253)
15:00.16btrachtagain could it be because it's an international call?
15:00.36[TK]D-FenderShouldn't be
15:00.39btrachti don't have international calling enabled
15:00.44[TK]D-FenderNo.
15:00.51[TK]D-FenderThis has nothng to do with ROUTES
15:01.50btrachti don't understand what in the trunk matches
15:02.04btrachtor doesn't match
15:02.43jmetrobtracht: i think he means to check authentication with the trunk [your provider]
15:03.22[TK]D-FenderAnd look at the CALL
15:03.38igcewielingbtracht: did you add the two options I suggested.
15:03.44btrachti did
15:04.08igcewielingbtracht:  then the issue should be resolved.
15:04.21igcewielingbtracht: you are running a GUI, correct?
15:04.45btrachti will check with my provider but my concern is that the CDR says the call was answered
15:05.06igcewielingbtracht: with the options I gave you, you should not get any more calls like that.
15:05.40btrachtok, thank you
15:06.17igcewielingbtracht: not having those options HIDES the real issue.
15:06.45btrachti would like to know what the real issue is
15:07.19[TK]D-FenderThe call is not matching yoru trunk.
15:07.27[TK]D-FenderThis is not a difficult concept
15:08.02btrachtif it's not matching the trunk why does the CDR say it was answered
15:08.26GreenlightGoing around in circles here aint we
15:09.23[TK]D-Fender[11:08]btrachtif it's not matching the trunk why does the CDR say it was answered <- it was accepted as a UNAUTHED CALL.
15:09.27[TK]D-Fenderbrtjsut like it said
15:09.57[TK]D-FenderAnd is making a shit job out of viewing your CDRs
15:19.45*** join/#asterisk jetlag (~jetlag@pool-71-168-192-247.cmdnnj.east.verizon.net)
15:21.11*** join/#asterisk Captain_Proton (~quassel@173.162.32.1)
15:24.26*** join/#asterisk k610 (~K610@cred.epid.ucl.ac.be)
15:24.29Captain_Protonneed some help I upgraded to Asterisk 1.8.23.0 from the repo's late night. every with asterisk is fine but AST does not show any of the ext. only thing I see on restart is this ERROR[1266]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL
15:24.56Captain_Protoncould this be causing it
15:26.19*** join/#asterisk cneb3000 (~cneb3000@2.221.241.85)
15:28.04igcewielingI recommend talking to the person who created the RPM
15:30.59*** join/#asterisk zerick (~eocrospom@190.187.21.53)
15:45.06Captain_Protonigcewieling: where do I find that out? it is just asteriskNow
15:45.46*** join/#asterisk War_Bear (~War_Bear@warbear.co.uk)
15:46.06igcewielingCaptain_Proton: no idea.  Maybe try a channel relevant the software you are using, like #AsteriskNow
15:46.34igcewieling~asterisknow
15:46.35infoboti guess asterisknow is based on Asterisk, but is difficult to support in #asterisk for a number of reasons.  Please seek support in #asterisknow instead.
15:47.36*** join/#asterisk Rumbles (~Rumbles@77.107.183.253)
15:47.59Captain_Protonthanks
15:48.21*** part/#asterisk Captain_Proton (~quassel@173.162.32.1)
15:50.23*** join/#asterisk ybeddyj (~ybeddyj@72.252.160.99)
15:51.14vlad_starkovQuestion: Anyone know how to use ParkAndAnnounce() application?
15:51.53*** join/#asterisk War_Bear (~War_Bear@warbear.co.uk)
15:53.17[TK]D-Fendervlad_starkov: https://wiki.asterisk.org/wiki/display/AST/Application_ParkAndAnnounce
15:53.33[TK]D-Fendervlad_starkov: Instructions seem pretty clear... what part of that are you having trouble with?
15:54.14vlad_starkov[TK]D-Fender: "dial" and "return_context" are not clear for me
15:54.29[TK]D-Fendervlad_starkov: dial - The app_dial style resource to call to make the announcement. Console/dsp calls the console.
15:54.40[TK]D-Fendervlad_starkov: ei the same thing you'd give DIAL()
15:55.38[TK]D-Fendervlad_starkov: return_context - The goto-style label to jump the call back into after timeout. Default priority+1. <-- just like it says.. when the app EXITS... here's where it goes.. unless you tell it to go to ANOTHER context
16:00.09vlad_starkov[TK]D-Fender: how to understand "dial"? Is it the location where the parked call will go to on timeout ?
16:00.41[TK]D-Fendervlad_starkov: No... it what it will DIAL... to ANNOUNCE to.
16:01.11[TK]D-Fendervlad_starkov: You are announcing to SOMETHING ELSE... not the transferer
16:01.20vlad_starkov[TK]D-Fender: It looks different than builtin parking feature
16:01.20*** join/#asterisk troyt (~troyt@2001:1938:240:2000::3)
16:01.47*** join/#asterisk [404] (~404]@12.179.117.114)
16:01.53[TK]D-Fendervlad_starkov: Park() tells the lot to the person doing the attended transfer to parking.  Parkandannounce .... DIALS what you tell it to and tells THAT the lot instead
16:01.56vlad_starkov[TK]D-Fender: why I'd need to announce to someone else?
16:02.10[TK]D-Fendervlad_starkov: Becausew you want to?
16:02.29vlad_starkov[TK]D-Fender: Oh, probably I need another app – Park()
16:02.34[TK]D-Fendervlad_starkov: can you not imagine any possible use for this?
16:03.48vlad_starkov[TK]D-Fender: hmm, Park() – Park yourself. <-- this makes me confused
16:04.07jmetromake the caller park itself to wait on hold?
16:04.09[TK]D-Fendervlad_starkov: I already told you how you use it...
16:04.45[TK]D-Fendervlad_starkov: ATTENDED TRANFER... YOU get the lot # read to you ... then you FINISH the attended transfer
16:04.59vlad_starkov[TK]D-Fender: what I need to do is to transfer the callee to *100 and to park him and to get parked slot number said back to me
16:05.19jmetropark by default will tell you the slot
16:05.33[TK]D-FenderWhich is what I said.  Twice
16:06.35vlad_starkovjmetro: I already tried default park functionality, but it turned out that I need to create my own as the default one does not provide the required functionality
16:06.52[TK]D-Fendervlad_starkov: It does everthing you've described so far...
16:06.58[TK]D-Fendervlad_starkov: So what are you not telling us?
16:08.39vlad_starkov[TK]D-Fender: The default park functionality works just great! One extra thing I need is to retrieve callerid information for explicit parked call.
16:09.02[TK]D-FenderClarify "explicit"
16:09.04vlad_starkov*before that call is being picked up
16:09.24[TK]D-FenderPick it up where?  How?
16:11.56vlad_starkovSay we have parked call on slot 704. We have special [parkedcalls_pickup] context that makes some logic with DB and then Dial(Local/704@parkedcalls,60,THK). So I need to retrieve the 704's callerid to save it in DB before I actually pick it up.
16:12.33*** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com)
16:12.51[TK]D-FenderThen go do it
16:13.04jmetroi told him that phones can do it
16:13.10jmetromy aastra has an xmlapp on it that does it.
16:13.12[TK]D-FenderYou can do whatever you want to scan your open channels and see the lot it's in and grab the callerID from that channel
16:13.19[TK]D-FenderThis has nothing to do with actual "parking"
16:13.49[TK]D-FenderAnd you should know what you're capable of doing via dialplan apps to access vars from other channels, etc.  This may require using AMI to get that info
16:13.51vlad_starkov[TK]D-Fender: sounds interesting, what should be my strategy here?
16:14.15[TK]D-Fenderlook at the dialplan apps & functions that can access other channels info.
16:14.24[TK]D-Fenderand then if you don't see it there.. AMI <----------
16:14.25vlad_starkov[TK]D-Fender: It is required to not use AMI.
16:14.45[TK]D-FenderThat may be your only choice
16:15.00[TK]D-FenderYou expect to pull some value out from ANOTHER channel
16:15.05*** join/#asterisk Leddy (leddy@krypton.evosurge.com)
16:15.13[TK]D-Fenderone channel does not normall get to spy on others that way
16:15.17[TK]D-Fenderthat is what AMI is for
16:16.27vlad_starkov[TK]D-Fender: I know that AMI could be solution, as it emits Park events.
16:16.28*** join/#asterisk jsjc (~Adium@238.Red-79-146-120.dynamicIP.rima-tde.net)
16:17.29GreenlightCan you store the CallerID, *before* parking the call ?
16:17.51vlad_starkov[TK]D-Fender: But it is required for me to not use AMI in this task, so that I decided to build my own call parking logic
16:18.03*** join/#asterisk Rumbles (~Rumbles@77.107.183.253)
16:18.12vlad_starkovGreenlight: Yes
16:18.35Greenlightvlad_starkov: So, do that then.
16:18.50[TK]D-Fendervlad_starkov: Well if you want to handcuff yourself... best of luck with that.  You should already see what little is offered.  If the callerID of that channel isn't part of it (and I seriously doubt there is another other way) ... best of luck to you.
16:19.24*** join/#asterisk threesome (~threesome@ip-94-113-12-233.net.upcbroadband.cz)
16:26.19jmetroregular parking definitely allows you to pull caller ID off the channel/lot
16:26.37jmetrooperator panels dont re-write parking
16:27.04[TK]D-FenderOperator panels.... USE AMI
16:27.19[TK]D-Fender</big_print>
16:28.14*** join/#asterisk nantou (~phonetic@gateway/tor-sasl/martinphone)
16:28.42jmetro^^^
16:28.44jmetro=)
16:28.56jmetro[that was my point]
16:29.15vlad_starkovGreenlight: [TK]D-Fender: Ok, I tried it with Park() and it works. And now the question is how to get the parking slot that the call was placed on?
16:30.02GreenlightYou've set yourself the challenge of not using AMI...
16:30.17[TK]D-Fendervlad_starkov: Go look at the channel and any extra AMI events thrown off
16:30.18vlad_starkovThere is the ability to explicitly set parking slot
16:30.54*** join/#asterisk sawgood (~sawgood@unaffiliated/sawgood)
16:31.24vlad_starkovGreenlight: that's because this server is standalone (no co-wrokers like adhearsion or so are allowed).
16:32.22WIMPyAnd how is that related to *YOU* not using AMI?
16:32.27GreenlightI don't understand the connectionm
16:32.31GreenlightIndeed
16:33.08vlad_starkovProbably I misunderstand something, How is it possible to listen to AMI inside dialplan?
16:33.12GreenlightIt's like coming in here and saying you can get SIP working, and the box isn't allowed to be connected to the network, and people must help you find a way around it.
16:33.44WIMPyDo you know what AMI is?
16:33.59[TK]D-FenderClearly not properly
16:34.01vlad_starkovWIMPy: manager interface, that I can connect remotly
16:34.13WIMPy"remote"
16:34.21vlad_starkovWIMPy: sorry
16:34.53vlad_starkovI'm not native english speaker :-)
16:35.34WIMPyWell, it's IP so it is remote in a sense. But I've never used it on anything but 127.0.0.1.
16:35.35GreenlightSo, you don't have a need for a remote user to manage a parked call ?
16:35.58[TK]D-FenderNobody reads anymore...
16:36.00vlad_starkovGreenlight: nope
16:36.03[TK]D-Fenderheads off to lunch
16:36.54GreenlightI thought you were trying to get the details of a parked call?
16:37.32vlad_starkovI have the strong terms to implement that logic only using asterisk facilities.
16:37.56WIMPyAMI is aterisks strongest tool.
16:38.06GreenlightWith AMI you can almost do *anything*
16:38.53vlad_starkovAm I right thinking that to use AMI I need some additional app(daemon) running on the same server or on remote server?
16:39.12WIMPyThe one you write.
16:39.40igcewielingAMI or dialplan, you're going to have to write code
16:40.21*** join/#asterisk cmendes0101 (~cmendes01@72.1.46.254)
16:40.50vlad_starkovWIMPy: that is restrictions. I only allowed to use asterisk facilities to implement it. No additional daemons and open ports. That is what I'm trying to say.
16:41.11GreenlightYour restriction, not asterisk's or ours
16:41.35WIMPySo you're not allowed to use "asterisk -r", either?
16:41.43vlad_starkovGreenlight: So I just asked is it possible to implement it without using AMI.
16:41.49GreenlightI wonder if you're allowed to have 5060 open?
16:41.52vlad_starkovWIMPy: that's allowed.
16:42.15WIMPyAnd why is one "remote" connection allowed and not the other?
16:42.19igcewielingvlad_starkov: AMI can be limited to 127.0.0.1 so you don't need open ports accessable from the network to run AMI.
16:42.25WIMPyI hope you get paid by the hour, BTW.
16:42.25GreenlightAMI *is* an asterisk facility
16:42.46WIMPyThat's what I said.
16:42.46GreenlightOk, what *exactly* is it you're trying to do ?
16:43.40vlad_starkovGreenlight: the goal is to get parked call's callerid just before the user pick that call up.
16:44.03WIMPyAnd what are you trying to do with that caller ID?
16:44.16Greenlight*before* the user picks up?
16:44.30GreenlightHow is the user "picking up" the call ?
16:45.16vlad_starkovGreenlight: yes, just before Dial(Local/${EXTEN}@parkedcalls,60,THK) will be executed.
16:45.45vlad_starkovAll the parked calls live in [parkedcalls] context.
16:45.46Greenlight${EXTEN} being the parking slot ?
16:45.53vlad_starkovCorrect
16:45.57GreenlightSo, use a global variable
16:46.01*** join/#asterisk ChadAragorn (~ChadArago@63-235-131-194.dia.static.qwest.net)
16:46.04GreenlightHave one for each parking slot
16:46.06WIMPyWhat do you plan to do with that caller ID?
16:48.03GreenlightOnly thing I could think would be Set(CONNECTEDLINE(number)=....
16:49.01vlad_starkovWIMPy: before Dial(Local/${EXTEN}@parkedcalls,60,THK) there is a sub that launches call recording, so I need to place the callerid of parked call in the recorded audio file name, like "call_from_102_to_1234567_(picked_up_from_708).mp3"
16:51.01vlad_starkovAhead of you question, I need this special sub as I make stereo call recording (caller on left and callee on the right).
16:51.11GreenlightWay I see it you have 3 options. 1. Use AMI. 2. Use a global variable. 3. Use the AstDB or another DB/AGI
16:51.48*** join/#asterisk danfromuk (~IceChat77@unaffiliated/danfromuk)
16:51.52vlad_starkovGreenlight: I chosen AstDB as a storage of parked calls information like callerid.
16:52.05GreenlightOk... so what's the issue ?
16:52.39GreenlightDon't you have access to the parking slot number ?
16:52.41danfromukHi, ive set Events: on with the AMI but I dont see when calls are placed on Hold. Is there any way to get that output to the AMI?
16:53.10Greenlightdanfromuk: "hold" is an odd one, sometimes it's just implemented by the endpoint
16:53.35danfromukGreenlight: I thought that but I can hear the asterisk onhold music.
16:53.50GreenlightIf that's the case you should see some events for it then
16:53.57jmetroyou should see it call the onhold music
16:54.03jmetrostarted onhold music class "company"
16:54.19*** join/#asterisk navaismo (~navaismo@189.191.233.140)
16:54.23GreenlightI *think* there are also "call" events that can be enabled
16:54.27*** join/#asterisk Pullphinger (~Pullphing@12.40.23.68)
16:54.27vlad_starkovNow, the issue is how to get parked slot number that the call has been parked on? I implement my own logic to park the call using Park() application. It parks calls well but if it return PARKEDAT?
16:54.28GreenlightIf memory serves
16:55.09danfromukGreenlight: you are correct. Not sure why it wasnt showing on the AMI, but it is now. Weird.
16:57.28*** join/#asterisk peetaur2 (~peter@x2f12083.dyn.telefonica.de)
16:57.37Greenlightvlad_starkov: You may need to manage the parking slot number yourself if you can't get it explicitly
16:57.49GreenlightPerhaps with some useage of the GROUP functions
16:58.55vlad_starkovGreenlight: That makes sense
16:59.22GreenlightI've always used AMI to provide this sort of functionality, so can't say that I'm too sure what gets set
16:59.44WIMPyGreenlight: Connectedline should happen automagically, I think.
17:00.18GreenlightIf that's the case, why not wait till *after* the calls connected and run the sub then
17:00.25GreenlightAnd grab the number out of CONNECTEDLINE
17:00.59WIMPyThat requires a current Asterisk version.
17:01.04vlad_starkovGreenlight: Probably in the future versions of asterisk it would make sense to add PARKEDAT variable to the Park() as it is in ParkAndAnnounce().
17:01.20Greenlightvlad_starkov: Or, you could juse use ParkAndAnnounce
17:01.23jmetro^
17:01.27jmetrotyped it before i could
17:01.33danfromukGreenlight: it appears that the AMI packets for OnHold are the same as UnHold. Any idea how I can tell the difference?
17:01.53GreenlightDon't they stay "on" and "off" or something ?
17:01.56Greenlight@ danfromuk
17:02.02WIMPydanfromuk: IIRC thera was a change to these events recently.
17:02.25danfromukGreenlight: There is a Musiconhold = yes and no. But is there always music?
17:02.27Greenlightvlad_starkov: Just looked at the docs - ParkAndAnnounce() would do exactly what you require
17:02.27vlad_starkovGreenlight: I started from ParkAndAnnounce() but as I figure out later this app is not exactly what I need, as it plays announcement to someone else.
17:02.35WIMPyBut I can't remember having had difficulties to distinguish them.
17:03.12Greenlightvlad_starkov: Doesn't matter about the announcement - just just need that "hook" to set the PARKED EXTENSION number to AstDB with the CALLERID
17:03.31danfromukWIMPy: when you say recently, was the change applied to version 1.8?
17:03.37GreenlightAs long as you can execute some dialplan and have access to the CALLERID *and* the PARKED EXTENSION, you're sorted
17:03.59WIMPydanfromuk: No, much later.
17:04.12danfromukOk, I'm stuck on 1.8 at the moment.
17:04.21Greenlighthttps://wiki.asterisk.org/wiki/display/AST/AMI+1.1+Changes <-- perhaps those
17:04.25WIMPy1.8 is the oldest version I want to think of.
17:04.34danfromukOk, if thats not possible, is there a channel variable that contains the hold status?
17:04.39GreenlightSpecificaly "For hold, there's a "Status: On" header, for unhold, status is off"
17:04.59GreenlightSo, looks like your version lacks those. That's annoying.
17:05.06WIMPyYou should get a channelstate, I think.
17:05.32jmetrovlad_starkov: parkandannounce plays audio and is suppressable
17:05.46danfromukOk, I'll try to find it. Or apply to have asterisk updated but not sure how much works involved in that yet.
17:05.54jmetrovlad_starkov: plays audio to the person putting the other person on park, suppress it by making a local channel do the park
17:05.56GreenlightDon't think there's an "hold" channelstate
17:07.04WIMPyNo, right, I only get a hold event.
17:07.43Greenlightdanfromuk: You *may* be able to patch your version by checking out the differences from current
17:07.53GreenlightIf you've some major issue preventing upgrade
17:08.12vlad_starkovjmetro: ParkAndAnnounce(PARKED,300,dial,return_context) <-- 1. what should be "dial"? 2. what extension will be called in "return_context"?
17:08.16GreenlightI wouldn't imagine there's anything crazy going on, other than an additional line for the event
17:08.30WIMPyBut that changes file says it changed from two different events to one with a status cariable. Both should do what you need.
17:08.32vlad_starkovOh, "return_context - The goto-style label to jump the call back into after timeout."
17:08.48jmetrovlad_starkov: yep, google it
17:09.44WIMPyErrm
17:10.25jmetroalso you can actually edit out the announce in the source code. Lol
17:10.33WIMPyDo you have to supply a supported version number to get the new format?
17:11.18WIMPyOr change the channels language to one that only contains empty files.
17:11.42jmetrothat might be messy.
17:12.30vlad_starkovjmetro: So far the best solution I found is just to put Console/dsp as "dial" argument.
17:13.06[TK]D-Fendervlad_starkov: And what is that doing for you?
17:13.29paulcAnyone used custom device states and Cisco SPA508G phones with BLF+SD to give indication of active/inactive features? I've got it working but have a question about the subscribe target versus speed dial destination (it doesn't seem to work if they're not the same?)
17:14.36vlad_starkov[TK]D-Fender: ParkAndAnnounce() requires to set "dial" attribute, I don't need it, but as in mandatory it should be stubbed with something.
17:14.52GreenlightDO make a dummy extension that does nothing but set your data that you need.
17:15.16GreenlightPresumable ${PARKEDAT} is set.
17:15.26vlad_starkovGreenlight: Hmm, makes sense
17:16.08[TK]D-Fendervlad_starkov: If you aren't using the Dial .... THEN YOU ARE USING THE WRONG APP
17:16.20[TK]D-Fendervlad_starkov: The entire point of that app is TO ANNOUNCE IT.
17:16.21Greenlight[TK
17:16.30jmetroParkAndAnnounce is the right app because it sets variables
17:16.33Greenlight[TK]D-Fender: He needs to set some stuff in the AstDB
17:16.43GreenlightPark doesn't seem to allow that to be done.
17:16.45[TK]D-Fendervlad_starkov: The only other thing to pass is the failover exten which Park() has options for anyway
17:16.51GreenlightUnless I'm mistaken.
17:17.16GreenlightEg, he needs store the association between the parking slot number AND the CallerID.
17:17.17[TK]D-FenderGreenlight: And where is that happening?
17:17.37GreenlightWell it's a bit of a hack, but the plan was to set it inside Dial
17:17.43GreenlightOf ParkAndAnnounce
17:17.55GreenlightAnd make use of ${PARKEDAT} which gets set
17:18.12GreenlightSo, he'll need Local/test@setmyparkedstuff
17:18.30GreenlightAnd inside that context he'll (in theory) have BOTH CALLERID and PARKEDAT
17:18.44GreenlightSo he can then set those in the AstDB
17:18.52*** join/#asterisk Ice_Strike (~Ice_Black@host213-120-117-212.in-addr.btopenworld.com)
17:19.07vlad_starkovGreenlight: that's correct
17:19.07Ice_StrikeGSM Gateway - Is it possible to change IMEI?
17:19.15GreenlightIce_Strike; No.
17:19.23GreenlightThat's illgal prison time stuff iirc
17:19.38GreenlightThey seriously frown on it
17:19.41WIMPyIce_Strike: Check teh gateways software.
17:19.43vlad_starkovIce_Strike: If I'm not mistaken, GoIP GSM gateway allows do some stuff.
17:20.13GreenlightI was told it was uber illegal...
17:20.18Ice_StrikeIs it?
17:20.26GreenlightUnless I'm thinking of something else
17:20.37WIMPyAFAIk noone cares about the IMEI.
17:20.49GreenlightIsn't that the unique identified for the SIM card ?
17:20.50Ice_StrikeIf gateway allows IMEI change - how can it be done via asterisk?
17:21.01WIMPyNo, that's IMSI.
17:21.05GreenlightAhh...
17:21.13GreenlightBloody acronyms
17:21.23WIMPyQuite definitely not. How could Asterisk do it?
17:21.39Ice_StrikeWe can change CallerID
17:21.44WIMPy[S]ubscriber / [E]quipment
17:22.17Greenlight^^
17:22.27WIMPyUse something else that can do it as gateway.
17:22.42Ice_StrikeWhat do you mean, such as?
17:22.56WIMPyosmocom
17:23.15WIMPyHow do you change caller ID?
17:23.22Ice_StrikeAh didnt know that
17:23.30Ice_StrikeWIMPy I don't know if it can?
17:23.30WIMPyAFAIK that's not even possible to send.
17:23.43Ice_StrikeI know we can via VOIP SIP.
17:24.00GreenlightYou trying to send differnt callerid's using a GSM gateway ?
17:24.11WIMPyYes, ITSPs and caller ID is an interesting topic.
17:24.17Ice_StrikeEg: exten => _X.,1,Set(CALLERID(num)=xxx)
17:24.30Ice_StrikeGreenlight Is it possible?
17:24.32GreenlightWe could never get that to work with GSM gateways
17:24.37WIMPyGreenlight: Unless I'm missing something, definitely no.
17:24.40GreenlightWas one of the reasons we ditched them
17:24.44WIMPyAFAIK that's not even possible to send.
17:24.58GreenlightThat's the conclusion I came to at the time.
17:25.11GreenlightWas glad to see the back of them tbh
17:25.41GreenlightScripts to keep track of inclusive minutes usage per SIM and whatnot.
17:25.47Ice_StrikeSomone manage to do but he wont tell me how.. he want £££££ lol
17:25.50GreenlightHorrible tech
17:26.14GreenlightIce_Strike: Take that with a grain of salt...
17:26.41Ice_StrikeHe said change the IMEI every call in order to change the caller ID.
17:26.43WIMPyWell, it is easy. Just hack someone elses SIM and use his account. But otherwise???
17:26.44Ice_StrikeThat it.
17:26.55WIMPyThat makes no sense.
17:27.09GreenlightBut you'd be using SOMEONE ELSES'S SIM
17:27.14GreenlightAs far as the network is concerned
17:27.20GreenlightThta's um....
17:27.22WIMPyCaller IDs are not set depending on the phone used.
17:27.27vlad_starkovGreenlight: One more thing that is not clear for me is how to set parking_lot_name for ParkAndAnnounce() ?
17:27.29Ice_StrikeActually.
17:27.45GreenlightHe does mean IMSI...
17:27.51WIMPyYou have to change IMSI.
17:27.54WIMPyYes
17:28.06GreenlightIce_Strike: Seriously... that's proper dodgy
17:28.15Ice_StrikeIs it?
17:28.18Greenlightvlad_starkov: Use the default, no ?
17:28.32GreenlightIce_Strike: It scares me that you need to ask..
17:28.35*** join/#asterisk w9sh (~sph@50-79-224-193-static.hfc.comcastbusiness.net)
17:28.43WIMPyWell, change IMSI usually means change to one of the many SIMs you have inserted.
17:28.58vlad_starkovGreenlight: there is multiple parking lots in the system, it needs to set parking lot explicitly.
17:29.06Greenlightvlad_starkov: So, choose one.
17:29.10vlad_starkovGreenlight: the documentation is not clear on it.
17:29.39GreenlightIf the SIM's inserted in the system, then you can choose to just use that SIM...
17:29.47Ice_StrikeOhh right.
17:29.49GreenlightWhy bother with IMSI's ...
17:29.53Ice_StrikeTrue.
17:29.55WIMPyYes, or in other terms change IMSI.
17:29.55vlad_starkovGreenlight: You mean I should set CHANNEL(parkinglot)?
17:30.14GreenlightI *suppose* if you have 1000 SIM's and only 10 slots in the GSM gateway, you could do it.
17:30.30WIMPyYou can also change IMSI by physically removing the SIM and insert another.
17:30.30GreenlightBUt I was always told that it was a very dodgy thing to play around with
17:30.47Ice_StrikeWhy GSM gateway exist?
17:31.03GreenlightUsed to be a cost thing for a lot of companies
17:31.05WIMPyTo make cheaper calls?
17:31.15GreenlightNowadays SIP is a lot cheaper and more accessible
17:31.32vlad_starkovGreenlight: CHANNEL(parkinglot) made the trick
17:31.37GreenlightBack 10 years ago, mobiles calls were a lot more expensive, and GSM gateways gave a nice alternative
17:31.45WIMPySo you are one of the guys who say SIP is cheaper?
17:31.45Greenlightvlad_starkov: Good :)
17:31.50Ice_StrikeWIMPy no, I meant Greenlight is implying GSM gateway is illegal
17:31.58Ice_Strikeor maybe missunderstood him
17:32.03WIMPywhat?
17:32.12GreenlightIce_Strike: No, GSM gateway isn't. Altering your IMSI is.
17:32.16Ice_StrikeOhhh
17:32.34WIMPyHe, everyone does so from time to time.
17:32.38GreenlightWIMPy: Your one of the haters when we sell SIP on price ? :P
17:32.51vlad_starkovGreenlight: [TK]D-Fender: Thank you guys and everyone for your help!
17:32.53WIMPyBut it's propper illecal to use someone elses IMSI, off course.
17:33.00*** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl)
17:33.09GreenlightThat's the point I was making.
17:33.12Ice_StrikeI am no plan to change IMSI
17:33.18Ice_StrikeThat is not my intention
17:33.20WIMPyI haven't seen cheap SIP, yet.
17:33.51GreenlightWIMPy: Cheap is relative. I'm comparing to similar volume over traditional PSTN networks.
17:33.53vlad_starkovGoodbye!
17:33.55vlad_starkov)
17:34.05Ice_StrikeFor personal voip - i use voip buster
17:34.06WIMPyGreenlight: So do I.
17:34.09Ice_Strikemuch cheaper than skype!
17:34.27GreenlightWIMPy: And you find, say ISDN30, cheaper than SIP ?
17:34.41WIMPyDepends on the number of channels you need.
17:34.49Ice_StrikeGreenlight By the way Lease Line installed other day :P
17:34.59WIMPyBut a PRI is much less than the same amount of IP bandwidth.
17:35.04GreenlightIce_Strike: Yay :)
17:35.05*** join/#asterisk jhlavacek (~jirka@87.89.218.63)
17:35.17Ice_StrikeOnly 10mbit upload and 10mbit download though
17:35.21Ice_Strikeenough for asterisk use only.
17:35.26GreenlightThat's plenty for your needs...
17:35.29Ice_StrikeYep
17:35.37Ice_StrikeShame Fibre not around this area :/
17:35.43Ice_StrikeWould be much cheaper if I go with that
17:35.45GreenlightWIMPy: Maybe different pricing over here in UK then...
17:36.08GreenlightIce_Strike: That's only pretend "fibre", and not suitable for your needs :)
17:36.38GreenlightAint you like 50+ seat call centre ?
17:37.48GreenlightRight am off - laters
17:44.51danfromukJust looking at the UPGRADE documentation. Am I correct in saying that an upgrade from 1.8 to 11 shouldn't be that difficult? I can't see anything regarding the dialplans that would need changing.
17:45.26WIMPyProbably.
17:45.43danfromukOk. I'll try it on a test server and see what the results are like.
17:45.44danfromukThanks
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19:19.49danfromukHi, I'm trying to upgrade from 1.8 to 11 and seeing the following when trying to Dial.  "No RTP engine was found. Do you have one loaded?"
19:20.04danfromukI've checked the build and the correct options seem to be selected. Any ideas what could be causing it?
19:20.27WIMPyIs it really built? It needs uuid now.
19:20.34WIMPy(since 11.5)
19:20.46WIMPyres_asterisk_rtp
19:21.12danfromukI've installed uuid and its marked as [*] in menuselect
19:21.27WIMPyNo autoload?
19:22.58danfromukstrange. just checked the modules directory and that module is missing.
19:23.27WIMPyDid you install to the same locations?
19:23.46danfromukjust used the default locations
19:25.36danfromukah. the /usr/lib folder is now /usr/lib64
19:26.04WIMPyo.O
19:27.18WIMPystill have them in /usr/lib/, even on the 64bit OS.
19:28.11danfromukexcept i was struggling to build and changed the path to lib64 as an attempt to build
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19:48.04igcewielingWow, porting my cell number to Vitelity was about as painless as it can get
19:48.57igcewielingdanfromuk: if you are just missing libasteriskssl or whatever when you try loading res_rtp_asterisk then all you need to is run ldconfig so asterisk will pick up the newly installed librart
19:49.01igcewielinglibrary
19:49.40jmetroigcewieling: did you just say "here's my number, port it for me" and boom it was there and configured?
19:49.53jmetroplus a plate of cookies?
19:50.16igcewielingjmetro: I had to fill out a form and upload a signature graphc.
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19:53.02igcewieling*sigh*  Upgraded to Asterisk 11 and FreePBX 2.11 and now our FreePBX backups are running once per min.
19:53.29jmetrorealtime backups!
19:57.37Vann:D
19:57.47Kattyhello my asterisk does not work at all how to fix plz?
19:57.52Kattyis urgent, srs answers only thx
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19:59.19jmetrokatty: your R&D division in Black Mesa should be able to provide resolution,
19:59.47Kattybooks trip to Arizona
19:59.55Kattygrabs sunblock and jets!
20:00.12jmetroDont forget to take a swing by Anomal..i mean Asterisk Materials.
20:00.22igcewielingjmetro: realtime running out of diskspace and breaking Asterisk
20:00.47jmetroigcewieling: that needs to be a quote.
20:02.15igcewieling"When Good Backups Go Bad: Next on Court TV!"
20:02.17jmetroOn a side note "Asterisk Realtime" is actually pretty good.
20:03.25igcewielingRealtime is like that girl/guy you see from a distance which looks really hot, but the closer you get the less hot they look.  Realtime is like that.
20:04.36jmetrobut then you get to know them and theyre into all the same things as you and they become way hotter than original.
20:05.14igcewielingjmetro: That is where the analogy breaks down.
20:06.02jmetrono :<
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20:14.42danfromukIssuing CORE SET VERBOSE 0 from the AMI doesn't seem to have an effect on the CLI in asterisk 11. Does anyone know whats changed?
20:15.07WIMPyIt is local to the shell now.
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20:15.40danfromukHow can I affect the CLI's verbosity level from the AMI so that when the AMI issues a DIALPLAN reload, it doesnt end up filling the CLI output?
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20:17.11WIMPyInteresting idea. Probably doesn't work any more.
20:17.46antonioXHi all
20:17.53antonioXI've a 2 fxs ports linksys ATA connected to an *. If someone calls from the PSTN and press 1, the call should be bridged to port 1 on the ATA, but I got a "302 Moved Temporarily" from the ATA: http://pastebin.com/raw.php?i=L06CK6jL
20:17.59danfromukHmmm. thats a problem. I record the CLI verbose output for later debugging if required. But I dont need to record all the output from dialplan reloads.
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20:18.14antonioXCalls from PSTN -> * -> port 2 works ok. And calls from port 2 phone to port 1 works ok. The only difference on the ATA config I saw between port 1 and 2 is the username/password and the port (5060 on port 1, 5061 on 2).
20:18.19antonioXextensions.conf: http://pastebin.com/raw.php?i=R0cyBLJn
20:18.20antonioXmacros.conf: http://pastebin.com/raw.php?i=qkVHY8Z3
20:18.29danfromukI used to set the verbose to 0 before doing a reload which worked fine.
20:18.33danfromukin 1.8
20:18.55antonioXWhat should I do to debug this?
20:19.07WIMPyNow you can have multiple shells open simultaneousely wioth different verbose levels.
20:19.25danfromukverbose
20:19.27danfromukoops
20:19.28danfromuksorry
20:19.39WIMPyOr debug
20:19.42Kattyi'll sorry your verbose in a minute.
20:19.50Katty(what?)
20:19.53Cuznerasterisk -vvvvvvvvvvvvvvvvvr or core set verbose 9000
20:20.04Cuznerover 9000 even
20:20.11jmetro9001 preferably
20:20.23Kattymake in the olden days, we didn't have verbose 9001
20:20.25jmetro[its impossible]
20:20.33Kattywe walked up hill, both ways.
20:20.37KattyIN THE SNOW.
20:22.06danfromukWIMPy: ok, looks like theres no way to resolve that.
20:24.42danfromukWIMPy: any idea if its possible to change the verbose level of dialplan reloads so they are higher than originally designed?
20:25.37Kobazsooooo umm
20:25.41Kobazi have a fun question for you guys
20:25.50jmetroBall-pit.
20:26.08Kobazon many calls but not all of them... on this one system they get three beeps in a row, it sounds like a truck backing up
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20:26.57danfromukWhen I had an issue like that, it turned out that the calls were being diverted through another PBX at the client's office. They were diverting calls but didn't tell me.
20:27.25Kobazcould that be like some sort of external call recording beeps or something
20:27.5544UAAB9Q3I am revising an issue from years ago.  Doing an AllPage, in the past, when there were -many- sip endpoints, the all page would fail
20:28.0344UAAB9Q3due to the command being "too long".
20:28.05danfromukCould be anything. I dont think asterisk plays beeps without showing it in the CLI
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20:28.09Kobazyeah
20:28.1144UAAB9Q3is this still an issue?
20:28.13Kobazdanfromuk: that's what i was thinking
20:28.24Kobazdanfromuk: and i don't have *any* polycom programming to do anything like that
20:28.47danfromukIs it only happening with incoming calls?
20:28.52Kobazyeah
20:28.54Kobazonly inbound
20:29.20danfromukThen I would say that the calls aren't coming in directly. They must be going through another system which is adding the beeps.
20:30.29WIMPydanfromuk: They already have multiple levels. You can always change the source...
20:31.05danfromukWIMPy: never changed the source before. and if I upgrade again, it will need redoing.
20:31.24danfromukplus, wouldnt know where to start looking.
20:31.39WIMPyOr you try to filter what you save.
20:31.58danfromuki just use logger.conf to save everything thats output to the cli
20:33.34WIMPyNot an wasy one.
20:34.02WIMPyYou could change the logger config instead of the verbosity.
20:35.57danfromukYeh, but it still needs to record other cli output for the call flows etc. Unless I can manipulate the logger from the AMI
20:36.26danfromukStill a bit annoying trying to use the CLI and having the dialplan automatically reloading every so often.
20:36.35WIMPyI guess you would have to change the config file and "logger reload".
20:36.53danfromuki think changing the source code is the simplest option.
20:37.04WIMPyProbably the safest.
20:37.25danfromukAny idea which file contains the code for dialplan loading?
20:38.20WIMPypbx_config would be my first guess.
20:39.45filepbx_config loads and parses extensions.conf, with it using defined APIs in pbx.c to manipulate the internal PBX core
20:40.12Qwellfile: go get on a plane
20:40.21fileI can not.
20:40.31filethere is a maintenance delay of unknown length
20:40.33QwellYou should try
20:40.35Qwelleep
20:40.51danfromuk@file, thanks. It is easy to spot where the verbose level is specified for outputs that are generated when reloading the dialplan?
20:41.08fileshould be
20:41.2344UAAB9Q3is there a better way of doing all page than we were doing back in the 1.4 and 1.6 days with an allpage.agi ?
20:41.3144UAAB9Q3using 1.8 or 10/11 now?
20:41.41danfromuk@file.  Ok, thanks
20:42.1144UAAB9Q3dunno how the hell my nick is "44UAAB9Q3"  Geesh.  should be "pigpen"
20:42.28igcewieling44UAAB9Q3: Try a channel for your GUI?
20:43.0344UAAB9Q3lets see if I can clear this up.
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20:44.08pigpenbetter?
20:44.15pigpenYes.  back to my pigpen self.
20:44.48pigpenanyway, I see multicast rtp in 1.8, dunno how that works with the poly's.  I know cyberdata has options for it.
20:45.30igcewielingpigpen: from my understanding the polycoms do the multicast all on their own, without involving Asterisk
20:46.17pigpenwell, years ago, I ran across an issue, when all-paging more than 30 phones (ie: I was tring from 30 - 300 sip devices), that the command length was too long
20:46.19pigpenand it failed.
20:46.35pigpenso I had to chop them up into smaller batches and call the batches in a meetme request.
20:46.38pigpenodd, pita.
20:46.58pigpenI'll look into the multicast way, could be nice.
20:50.39igcewielingpigpen: read the polycom documentation before anything else.
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20:56.37pigpenyears ago.
20:56.46pigpensurly it hasn't changed.  ;-)
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21:00.39jmetroIf i allow ulaw,g722 and the connection has ulaw,g722 which one does it choose
21:02.31[TK]D-Fenderulaw
21:03.05jmetroand if i reverse that?
21:03.12jmetrog722,ulaw on both
21:04.18[TK]D-Fenderg722
21:04.39leifmadsenjmetro: order matters
21:04.44leifmadsenfirst ordered is highest priority
21:04.51leifmadsenif supported, will use that
21:05.00leifmadsene.g. first codec listed is preferred
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21:05.24jmetroright, now that i think about it
21:05.39jmetromakes more sense as a list rather than a pool
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21:33.42guldgossenhello :)
21:34.06WIMPyA golden what?
21:34.40pigpenegg
21:34.44guldgossenhmm..boy..or something :p
21:34.54psandersAny agi/ami dev in this channel? I need some help to validate the RC1 of my framework (http://astivetoolkit.org).
21:35.01WIMPyoh
21:35.04WIMPyok
21:35.42WIMPypsanders: And what exactely is your question?
21:35.52pigpenhmm..I am kinda disappointed now.
21:38.17psandersWIMpy: I'm asking for people with experience with agi/ami willing to help testing this software.
21:38.51WIMPypsanders: Aren't that those people who don't need it?
21:40.32psandersWIMPy: Not sure I understand your question.
21:40.36guldgossenI have a question about asterisk
21:41.01guldgossenwhat does failed mean under disposition in the cdr log?
21:41.09pigpenguldgossen, just one?  shit, I have been using/developing on it for years and I probably have 10 a day.
21:41.22guldgossenhehe
21:41.37WIMPypsanders: The stuff you did there seems only of interest to those who don;t have the knowledge to check for its sanity.
21:41.46guldgossenoh..i should say freepbx btw
21:41.59guldgossenfreepbx/asterisk :)
21:45.33psandersWIMPy: You may be right...
21:46.29guldgossenanyone knows?
21:46.58guldgossenseems like I only get it when making calls using my vpn server
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21:48.37WIMPyIntereesting. I just got two calls at about the same time, but they didn't get to any phone before one of them gave up.
21:51.58navaismoguldgossen, failed means failed --captain obvious attack again- so cant contact the peer the trunk or so...
21:57.02guldgossennavaismo: ok..thanks
21:57.23WIMPyI have an issue with Asterisk answering a MESSAGE with "415 Unsupported Media Type" which then unfortunatly kills the call. Is there anything I can do to change that behaviour?
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21:57.45WIMPyI see there is soem MESSAGE related configuration, but I think that only for SIMPLE?
22:00.49WIMPyOr rather only for out-of-call messages, which are not my concern.
22:02.45psandersWIMPy: Googleit -> https://issues.asterisk.org/jira/browse/ASTERISK-15802
22:03.09WIMPyI see that I'm not alone, But I haven;t found the answer, yet.
22:03.29psandersWIMPy: One of the 2 SIP endpoints does not have a G.729(or any other codec) license
22:03.57WIMPyDid you read my question? It's about MESSAGE messages.
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22:05.44psandersWIMPy: Yes.
22:06.11WIMPySo no CODECs involved here.
22:06.40psandersWIMPy: Ok...
22:08.16WIMPyGoogle gives me several identical questions, but the only extremely dirty suggestion so far was to tchange the error message in the source to "202 Accepted".
22:10.11WIMPyis constantly amazed at how unusable all this stuff is in practice :-(
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22:21.28WIMPywonders if chan_gulp could help.
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22:56.29WIMPyOk, patch is available.
22:57.14WIMPywonders how much brainfuck it would be to use libpri from chan_sip.
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23:03.48igcewielingedges away from WIMPy and makes sure there is garlic on hand
23:04.32WIMPyI want a whole acre of garlic between me and SIP.
23:04.42WIMPy(at least)
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