IRC log for #asterisk on 20150718

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01:27.25ledoktregreets.  anyone on here pretty good with compiling sangoma wanpipe drivers?
01:29.08WIMPyFor me DAHDI is more than enough, so you have my commiseration.
01:30.18ledoktreThanks WIMPy.  My problem was that I upgrade my rig from Debian Wheezy to Jessie last night not ever dreaming I'd have an issue compiling the new driver.
01:30.50ledoktreTrying to figure out if its a kernel compatibility issue or if Im somehow missing something obvious but thus far I have had no success.  No one's been around the #sangoma channel either
01:31.46WIMPyDon't they tell you what the latest supported Linux is, either?
01:32.38ledoktreNo. I saw in the change log a previous version noted it supported 3.14 kernel I believe.  Unfortunately Im on 3.16.
01:32.46ledoktreDidnt see that till after the "upgrade"
01:34.41ledoktrewas hoping since there was a version or two released since then it included 3.16 support
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01:48.36moyledoktre: do you have the compilation log? put it on pastebin
01:57.52ledoktremoy -sure.  I'll do that right now.
01:58.44WIMPyJust to catch the train: Is it possible on Debian to deinstall the Kernel?
01:59.13ledoktremoy : stepping out for just a bit but here's my pastie : http://pastebin.com/3WLu3xZA
02:01.58WIMPyJepp. That looks incompatible. Although there's a slight mismatch between kernel and headers as well.
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02:27.07moyledoktre: yeah, changes in the kernel to the proc api broke the driver, I know we got it to compile on Arch with kernel 4.x, but I think we've not released that driver yet, let me check if we have a beta driver that I can share
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02:40.55*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.4.0 (2015/06/04), 11.18.0 (2015/06/04), 1.8.32.3 (2015/04/08); Standard: 12.8.2 (2015/04/08); DAHDI: DAHDI-linux 2.10.0 (2014/08/13), DAHDI-tools 2.10.0 (2014/08/13); libpri 1.4.15 (2014/06/16) -=- Asterisk wiki: wiki.asterisk.org -=- Code of Conduct: bit.ly/1hH6P22 -=- Logs: bit.ly/1s4AKKu
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02:48.33chandoohi
02:50.40moyledoktre: try this ftp://ftp.sangoma.com/linux/custom/7.0/wanpipe-7.0.14.1.tgz
02:51.29moyhas not been run through QA, but that one compiles on 4.x, so might work for your kernel too
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03:39.50chandoohi
03:40.16chandoowhen i dail outside number from asterisk i am getting below error
03:40.18chandooWARNING[101031][C-00000099]: chan_sip.c:23296 handle_response_invite: Received response: "Forbidden" from '"101" <sip:123456789@sip.phonepower.com>;
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04:56.13ledoktremoy: TY for the link.  I'll test it out.  How'd you know about that link?  I looked all over for a beta ;-P
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05:04.49ledoktremoy: Not out of the woods yet but it passed that spot.  Linking Wanpipe Driver and protocols right now… YAY!! TY!!
05:08.25ledoktremoy:  YESSSS!!! Install compiled fine.  Dont know where you dug up that link but THANK YOU !
05:11.15drmessanoledoktre, read this messages
05:11.28drmessano"let me check if we have a beta driver that I can share"
05:11.34drmessanoSounds like he works there :)
05:12.21ledoktreahhhhhh.  I wondered.  I dug around a bunch and could find nothing newer.  The beta on the ftp was from quite a while ago.    I just got back and I saw the link guess I didnt read up far enough.
05:12.27ledoktre:D :D :D
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10:45.16chandoogood morning
10:54.48chandoowhen i dail number to outside i am getting this message,, how to fix this
10:54.55chandoochan_sip.c:23296 handle_response_invite: Received response: "Forbidden" from '
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11:03.56WIMPyCall somewhere you're allowed.
11:04.08WIMPyOr check your username/password.
11:04.28chandooWIMPy, that is what i am trying
11:04.40chandooi am getting message forbidden
11:05.00chandooi can receive calls sucessfully between devices
11:05.18chandoobut not able to dail outside calls
11:05.48chandooalso not able to dail from outside to asterisk main line, it goes to VM
11:06.48chandoo<--- SIP read from UDP:206.15.150.6:12060 --->
11:06.49chandooSIP/2.0 403 Forbidden
11:06.49chandooVia: SIP/2.0/UDP 192.168.5.30:5060;received=74.89.97.20;branch=z9hG4bK04789a9b;rport=5060
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13:54.15joker_89hi
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13:54.55joker_89i have some doubts..
13:55.38joker_89for the outgoing calls i need voip?
13:57.03[TK]D-FenderNo
13:57.07[TK]D-FenderIt is an option though
13:58.16joker_89I'm a bit lost...
13:59.06joker_89If i have a conventional phone line can i use it for outgoing calls?
13:59.21joker_89and recive calls from a VOIP provider (with the same number)
13:59.45[TK]D-FenderThre are interfaces for regular analog lines, ISDN BRI & PRI lines, interfaces for GSM antennas, etc
13:59.51joker_89so,.. Do I need an hibrid phone (SIP and analog ports)
14:00.18[TK]D-FenderYouif you have a physical line you want to use, then you need an interface for it.
14:00.33[TK]D-FenderThis gets the call to Asterisk
14:00.53[TK]D-FenderWhat kind of phones you want to use is up to you.  it has no relationship to the interface you use for that line
14:00.54joker_89an interface like whcih one?
14:00.57[TK]D-FenderAsterisk sits in the middle
14:01.38joker_89an interface is a FOX Card?
14:02.31[TK]D-FenderFor analog lines there are things like PCI(e) cards, and there are also VoIP gateways which take in the analog line and it talks using a VoIP protocol to your server so you don't have to install a special card into a specific server you run Asterisk on
14:03.05[TK]D-FenderFXO interface is the interface to the line itself.  What that device uses to talk to your Asterisk machine will vary as I mentioned above
14:03.14[TK]D-FenderThere are actually FXO > USB as well
14:03.54joker_89which is cheaper? the VOIP gateway or PCI(e) cad?
14:03.59joker_89card
14:05.25Milencovoip is usually cheaper
14:05.45Milencoat least for outgoing calls
14:06.10[TK]D-Fenderjoker_89, He is referring to using an ITSP and not talking about using your actual line.
14:06.18joker_89can i call directly to one internal extension from outside?
14:06.48Milencoyes you can configure this in the dialplan
14:06.53[TK]D-Fenderjoker_89, When a call comes in from an analog line on some kind of interface an that call is sent to Asterisk you can do whatever you want with it
14:07.23joker_89but if my number is like 99112233
14:07.33joker_89then can i call directly to one extesion using 99112233001
14:07.34Milencowhat do you want to achieve?
14:07.57[TK]D-Fenderjoker_89, Not with analog.
14:08.20[TK]D-Fenderjoker_89, Analog noramlly doesn't get extra digits or anything like that.  Analog = dumb
14:08.35MilencoISDN does work like that
14:08.44[TK]D-Fenderjoker_89, the line is simply "ringing" and all you get on top of that is Caller ID normally.
14:09.00Milencoand so can VoIP, at least I have one provider that does it like that
14:09.29joker_89well, i want to have 2 phones in a new company (with currently 2 persons). Its a familiar new company. They want to receive calls from outside and then with a record voice someting like : "press 1 to talk with X department or press 2 to talk with Y department"
14:09.35[TK]D-FenderI've never seen an ITSP like that.  Normally they just sell fixed-digit DID's
14:09.52[TK]D-Fenderjoker_89, Simple auto-attendent.  sure
14:09.53joker_89also, they want to redirect calls to their mobilphones when they are away from the office
14:10.08[TK]D-Fenderjoker_89, Also simle.  This is all DIALPLAN
14:10.10MilencoCorrect [TK]D-Fender, most providers I know do it like that
14:10.21Milencobut we use a provider in Germany, called Dialoga, which offers us this option
14:10.40joker_89so, do you recomend to use all over VOIP?
14:10.45MilencoUnfortunately their service/support is lacking so now we are in a search for another provider offering this :(
14:10.48joker_89currentlly, i dont have any conventional line
14:11.03joker_89because the company is not set up right now
14:11.10joker_89"micro company"
14:11.17[TK]D-FenderMilenco, I've never heard of this service.  It's a very EU thing that I've only haerd reference to from those using BRI/PRI directly
14:11.28MilencoIndeed, this option I only know in Germany
14:11.33joker_89Me, i am from Spain
14:11.41[TK]D-FenderMilenco, Seems you found a special one...
14:11.47MilencoI manage servers in NL/BE/PL which don't know of this either
14:12.32[TK]D-Fenderjoker_89, If you don't have a physical line you want to use then you can easily shop around for an ITSP instead of paying for a physical line and having to buy an interface for it.
14:12.34joker_89guys, but what happens if you want to use VOIP in your mobile phone thorught 3G but your bandwitch is consumed and you only have 16kbps GSM
14:12.35joker_89?
14:12.36Milencoanyway joker_89, i would recommend this normally: If you have low call volume and stability is more important, stay with analog or digital (ISDN) line
14:12.45Milencoif you have a lot of call volume and price is important, go with voio
14:12.47Milencovoip
14:12.54MilencoAsterisk can handle all these kind of calls
14:13.21joker_89no.. i have low call volumne
14:13.25Milenco(provided you have hardware for it, like a PCI card for example to handle analog or ISDN calls)
14:13.47joker_89only 2,3 peoples for the next 2 years, not more
14:13.57[TK]D-Fenderjoker_89, with mobile, you're either using mobile data and VoIP on top to get to your server, or WiFi.  Normall calling from/to using GSM is through your provider and not related to your server really
14:14.00Milencodo you want multiple incoming numbers/
14:14.20joker_89at least 2 incoming numbers
14:14.21Milencoin that case go with ISDN and buy a ISDN card for your Asterisk server
14:14.27joker_89to talk simultaniosly
14:14.40[TK]D-FenderOr get 2 DID's with an ITSP
14:14.43Milencoof course you could do VoIP as well, but in that scenario i wouldn't recommend it
14:15.00[TK]D-FenderDepends on the final cost
14:15.12MilencoWill you configure the Asterisk server yourself joker_89?
14:15.24joker_89yes with asterisknow
14:15.43joker_89i never heard about ITSP in spain :S
14:16.12MilencoOke, no offense here but Asterisk itself can be a little bit tricky to grasp with the whole contexts and dialplan thing
14:16.25MilencoITSP is a US term mostly :P
14:16.36Milenco(I think :P, I havent heard of it either)
14:16.45joker_89it means ISP_
14:16.48joker_89?
14:16.53[TK]D-Fender~itsp
14:16.53infobot[~itsp] An ITSP is an Internet Telephony Service Provider (or VoIP telephone company). They allow you to either SEND calls to the PSTN (this is called termination), RECEIVE calls from the PSTN (called origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs.
14:16.58Milencoit means VoIP provider
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14:17.11joker_89ahh ok
14:17.14[TK]D-FenderYou talk VoIP to them and they send yor calls to the PSTN
14:17.19Milencoohw
14:17.29[TK]D-FenderThey have the hardware lines so you don't have to deal with that
14:18.24Milencook thanks :)
14:18.39joker_89RDSI = IDSN
14:19.10Milencoyes  RDSI = ISDN
14:19.23MilencoBRI allows 2 simultanous calls
14:19.30MilencoPRI allows 30 simultanious calls
14:20.04joker_89whats BRI?
14:20.14[TK]D-FenderITSP = allows bandwidth worth of simultaneous calls :)
14:20.17[TK]D-Fender~bri
14:20.17infobot[~bri] Basic Rate Interface - a form of ISDN that consists of 2 * 64kbit/s Bearer (B) channels and 1 * 16kbit/s signalling channel (D)
14:20.42[TK]D-Fendertypical residential version of ISDN
14:20.45joker_89ok, well now i am lost again. I dont know waht is the best for me
14:21.01Milencooke joker_89, check out this book: Asterisk: The Definitive Guide
14:21.29[TK]D-Fenderjoker_89, that's why yo should for ITSP's that can sell you DID's in the area you want them for, and compare their service prices to that of getting telco lines locally and having to buy  an interface for them
14:21.33joker_89but i dont know if I need VOIP and Asterisk for my case
14:21.51[TK]D-Fenderjoker_89, It isn't a question of "need" It's a question of how you WANT to do it
14:21.59[TK]D-Fenderjoker_89,  You have options
14:22.09[TK]D-Fenderjoker_89, Any of them can work.
14:22.34Milencojoker_89, to be quite honest it might be best to outsource this to another company
14:22.49Milencoand let them setup your phones and lines/numbers
14:24.32joker_89the only requisits are: posibility to use 2 lines at the same time. Call directly to a specific extension from outside. Call to the unique number and a voice whre you can redirect this call to one of the two phones. Posibility to redirect the phone to the mobile phone if one of the 2 persons is outside
14:25.09[TK]D-Fenderthen you need 3 phone numbers
14:25.22[TK]D-Fender2 for direct, one or the general that offers tha menu
14:25.56[TK]D-FenderHow many NUMBERS you have can be separate from how many CALLS you can support
14:26.00joker_89I thought that you can add some sufix to the general number
14:26.28joker_89to call to each individual extension
14:26.43[TK]D-FenderThat is a typically BRI/PRI concept that you'd need to get a line from your telco for
14:26.54[TK]D-FenderAnd an interface to match
14:27.23joker_89so, which are my options? :S
14:27.28[TK]D-Fenderthat's it
14:28.18[TK]D-FenderYou either get standalone DID's with an ITSP, or a BRI lines from a telco.  I suspect PRI would cost more than is worth it for you
14:29.35joker_89DID's?
14:29.41[TK]D-Fender~did
14:29.41infobotfrom memory, did is Direct Inward Dialing, or just a phone number
14:30.25[TK]D-FenderAKA forget "adding digits to a prefix"
14:38.13joker_89whats frevoipdeal ?
14:38.23joker_89freevoipdeal
14:38.42[TK]D-Fenderan ITSP
14:40.00joker_89in spain i heard that some people uses freevoipdeal for outgoing calls and anotehr provider for ingoing calls
14:40.20[TK]D-Fenderyou can have as many ways in and out as you want
14:41.22joker_89but with VOIP you need always to pay for each minute
14:41.36[TK]D-Fenderno
14:41.48joker_89compared with phone calls where you pay 30$ and you can do so many calls do you want
14:42.02[TK]D-Fenderthat depends on what kind of plans they offer
14:42.25[TK]D-FenderSome providers offer unlimited calls across particular regions, etc at a flat rate
14:42.29[TK]D-Fenderit DEPENDS
14:43.23[TK]D-Fendersame with incoming
14:44.02[TK]D-Fendereash ITSP has their own bundles of service.  They are not all alike.  You will actualy have to shop around
14:44.23joker_89well, incoming its free
14:44.30joker_89no?
14:44.55nbjoergit depends
14:47.07[TK]D-Fenderincoming isn't free witha  phsyical line
14:47.19[TK]D-FenderYothere is a base monthly fee that covers that part
14:47.30[TK]D-Fenderas well as a certain amount of outgoing
14:48.02[TK]D-Fenderwith an ITSP... they have to use a connection to the PSTN that THEY pay for... so it's not scommon to see them give you service for free....
14:52.54joker_89[[TK]D-Fender] and whats this solution? http://www.onedirect.es/productos/gigaset/mini-centralita-gigaset-c530-ip
14:53.26[TK]D-Fendertypically something that you wouldn't have Asterisk in the middle of at all
14:53.52[TK]D-Fenderwhen yuo wnt a dumb phone that talks direct to an analog line ... and can also send the call over VoIP.
14:54.19[TK]D-FenderIf you had an Asterisk server you would typically let Asterisk handle your phsical line and therefor NOT plug it into your phone.
14:54.39joker_89with the link that I put you can only receive one call?
14:54.40[TK]D-Fenderso this phone is clearly for those who want VoIP for cheaper calls to specific places but still want an analog line for local use
14:54.42joker_89sorry
14:55.15[TK]D-Fenderit talks VoIP AND Analog.  how it manages the handsets... depends on the unit itself
14:55.35[TK]D-Fender3 líneas independientes: 1 analógica + 2 líneas IP.
14:55.35[TK]D-Fender3 llamadas simultáneas
14:55.41joker_89you can make calls and receive calls uisng VOIP or Analog as you want?
14:55.46[TK]D-FenderWhich it does summarize for you right on the page...
14:55.55[TK]D-Fenderread it's manual to see how it works
14:57.24[TK]D-Fenderbut forget the idea of having a menu, and possibly separate voicemail boxes, routing calls back OUT like you wanted for redirect, etc
14:57.30[TK]D-Fenderthat thing is not a PBX
14:57.46[TK]D-Fenderit is meant for home users
14:58.11joker_89ok
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15:06.43joker_89[[TK]D-Fender] using that hibrid version, the incoming calls goes to one specific phone ?
15:06.50joker_89or all phones
15:07.42[TK]D-FenderDepends on the unit
15:07.54[TK]D-FenderWhen I said "read its manual" I meant it
15:08.08[TK]D-FenderI don't know if that is any smarter than I imagine it is dumb.
15:08.26[TK]D-Fenderbut that that price point... realize it is a consumer device and WILL be mostly dumb
15:12.10joker_89I think that its useful if you want to use conventional calls and also have the psobility to do VOIP calls
15:13.24[TK]D-FenderAs a boring consumer who has no menus, redirects or other funny business and want to read a 30 page manual at most to do it in 1 device
15:14.06*** join/#asterisk moe` (~moe@node-be028882.scarlet.cw)
15:14.13moe`sighs
15:14.32moe`guys, I still have not got asterisk to skype goodness
15:14.43*** join/#asterisk gusto (~gusto@2001:4c50:62f:ad13:8219:34ff:fecf:17f0)
15:15.24moe`I got the SIP registering on skype side, so it connects... the register statement in sip.conf works
15:15.36moe`but I am doing something horribly stupid
15:15.47moe`cannot get it to allow outbound calls
15:16.26[TK]D-FenderGo look at what you're sending and how they are responding then
15:16.46*** join/#asterisk gusto (~gusto@2001:4c50:62f:ad13:8219:34ff:fecf:17f0)
15:17.56*** join/#asterisk happy-dude (uid62780@gateway/web/irccloud.com/x-elwcgukhsbcxwhrr)
15:18.54*** join/#asterisk gusto (~gusto@2001:4c50:62f:ad13:8219:34ff:fecf:17f0)
15:20.27*** join/#asterisk gusto (~gusto@2001:4c50:62f:ad13:8219:34ff:fecf:17f0)
15:26.05moe`newbie here
15:26.15joker_89[[TK]D-Fender] if you redirect your call to your mobile phones, what happens if the GSM network is to low?
15:26.19moe`what's the CLI to show sip connections?
15:26.20[TK]D-Fender"sip set debug on"
15:27.10[TK]D-Fenderjoker_89, there is no "redirect" BTW, it's your server sitting in the middle.  You are taking a call in, then bridging it out.
15:27.18joker_89ok
15:27.26joker_89but what happens if my phone network is 16kbps?
15:27.31[TK]D-Fenderjoker_89, And what happens.. is the same thing you hear when you call it in that condition any other way you have ever called it
15:27.55[TK]D-FenderIf I take a dumb analog line... then call a mobile user... it sounds like what it sounds like
15:28.13moe`man, sip set debug on is verbose
15:28.36[TK]D-Fendermoe`, There is only 1 way.  You either see the packets or you don't
15:29.17moe`I know, but this whole SIP/PBX thing is new to me
15:29.35moe`I've written libpcap stuff that integrates with RRDtool, so I know about packets
15:29.42moe`but I'll be damned if I can get this going
15:29.46moe`<- fail
15:30.22[TK]D-Fenderthen change your approach
15:30.31joker_89i dont understand your response
15:30.40joker_89in spain the network is limited per month
15:30.46joker_89like 1Gb / month
15:30.54joker_89then is like 16kbps
15:31.08Milencojoker_89, you are doing everything at the same time
15:31.21Milencofirst pick your provider type, be it voip or isdn
15:31.25Milencothen compare providers
15:31.31Milencothen setup asterisk
15:31.58[TK]D-Fenderjoker_89, Are you referring to DATA over GSM?
15:32.03[TK]D-Fenderjoker_89, You aren't being clear
15:32.18[TK]D-Fenderbecause bitrates also apply to voice codecs
15:32.18moe`if I offered a URL link to my sip.conf and extensions.conf would someone care to look and tell me how stupid I am?
15:32.37[TK]D-Fendermoe`, configs won't say what's actually happening.
15:32.41[TK]D-Fendermoe`, the debug will
15:32.51moe`I hear that
15:32.58moe`I expect I have stupid misconfig though
15:33.04joker_89i mean DATA over GSM yes
15:33.17[TK]D-Fenderyup, and seeing the comms is the first line of attack
15:33.18joker_89if i dont have wifi in my mobile phone
15:33.38Milencoyou can also just bridge the call to a GSM number
15:33.55[TK]D-Fenderjoker_89, if you are planning on using DATA to call and your BW drops to that rate you are dead in the water
15:33.59Milencoso moe`, what are you trying to achieve?
15:34.21[TK]D-FenderMilenco, trying to get Sype's SIP seervice to accept his calls
15:34.26moe`trying to get asterisk to allow calls via the Skype Connect SIP connection
15:34.32[TK]D-FenderMilenco, we're just waiting ont he debug to see what's happening
15:34.39Milencooke, thanks :)
15:34.40moe`I can get "local" calls
15:34.51moe`so I can call other users extensions
15:34.55joker_89mmm
15:35.00moe`the skype register works
15:35.06joker_89but VOIP calls work with 16kbps BW?
15:35.17[TK]D-Fenderjoker_89, no.
15:35.19moe`at least, manager.skype.com shows there is a connect
15:35.22Milencono joker_89, it depends on the codec used
15:35.32[TK]D-FenderMilenco, No.  period.  not happening
15:35.34joker_89which is the typical codec
15:35.39[TK]D-Fenderjoker_89, NO
15:35.50Milencomaybe im misunderstanding as well :D
15:36.02Milenco(just trying to give back some support after receiving so many)
15:36.45Milencoso you want to do outbound calls via skype over asterisk moe`?
15:36.51Milencoor receive skype calls in asterisk
15:36.54moe`yeah, that's the idea
15:36.56moe`outbound
15:37.03moe`inbound was going to be step 2
15:37.29Milencooke, then have you got an entry with the sip-server/credentials/etc in your sip.conf?
15:37.41Milencoand a dial command to that SIP-channel in your contexts?
15:37.55moe`well this is where I fail
15:37.59moe`I believe
15:38.14Milencohave you used external sip providers before?
15:38.24moe`no I'm a newbie to SIP
15:38.36[TK]D-Fendermoe`, Show. Us. The. Failed. Call.
15:38.42moe`I have the register statement for skype and apparently that works
15:38.56Milencoif you use register => in sip.conf thats for incoming calls
15:38.57moe`[TK]D-Fender -- it's verbose
15:39.04[TK]D-Fendermoe`, I know.
15:39.16moe`Milenco: no way.  fuck me gently with a chainsaw
15:39.25moe`see, I just learned something
15:40.15Milencoto dial over a sip channel do something like Dial(SIP/yourreferencetosipconf,0123456789)
15:40.20Milencoto whatever number you like to dial
15:40.58Milencojust like you dial internally to SIP/123 or whatever, dial to SIP/skype and supply the number you want to call as an argument
15:41.23Milencofor example, i got [flowroute] in my sip.conf and below there all the settings
15:41.46[TK]D-Fender<Milenco> to dial over a sip channel do something like Dial(SIP/yourreferencetosipconf,0123456789) <- after you fix the syntax
15:41.51Milencoin the dialplan context flowroute is set to, i use it to dial a number
15:42.09Milencoindeed, [TK]D-Fender, too lazy to search for an exact sample
15:42.26Milencoits SIP/skype/012345 i think :P
15:43.08Milencoso yeah moe`, in that case the dialplan and sip config would help to solve your issue (without passwords etc.)
15:43.19moe`yeah lemme do that
15:43.32moe`hang on I'll sanitize the configs and post them on my site
15:44.24Milencobtw, check out the answer here moe`: http://serverfault.com/questions/370118/skype-connect-as-sip-trunk-for-asterisk
15:46.59moe`https://unixninja.net/files/sip.conf
15:47.05moe`https://unixninja.net/files/extensions.conf
15:47.13moe`laugh and tell me how stupid I am
15:47.43Milencoexten => _XXXXXXXXXXX,1,Dial(SIP/skype/${EXTEN})
15:47.46Milencodoesnt below in sip.conf
15:47.54Milencobelong*
15:48.01Milencobut in extensions.conf
15:48.20Milencoit should be uncommented there
15:48.35Milencoand notice it will only work with number that exact length
15:49.03Milencousually outbound numbers can differ in length and i usually use something like _XXXX. to grep them
15:49.15joker_89[[TK]D-Fender] which is the use of a GSM dongle for asterisk :S?
15:49.22Milencoso any 5 or more digits are triggered
15:49.22moe`so the exten statement is not sip.conf, but extensions.conf
15:49.28Milencoindeed
15:49.37joker_89for calls on GSM/3G netwrks
15:49.48Milencodo that and then show the verbose asterisk log for that call if it still doesnt work
15:49.56Milencoand 'sip show peers' should show skype connected
15:50.29*** join/#asterisk fornax (~fornax@85.183.43.221)
15:50.33moe`sip show peers does indeed show skype
15:50.40Milencojoker_89, so you can do calls from/to your gsm in asterisk
15:50.58MilencoStatus is OK too moe`?
15:51.36Milencojoker_89, to/from your sip card that is. as in, take and do calls from your gsm number
15:51.39moe`banshee*CLI> sip show peers like skype
15:51.39moe`Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
15:51.39moe`skype/99051000268278      63.209.144.201                              Yes        Yes            5060     Unmonitored
15:51.55moe`erg, I guess that was me being bad dog, sorry on the paste
15:52.08Milencolol
15:52.56joker_89wow
15:52.59joker_89taht sgreat Milenco
15:53.01Milenconot sure why its unmonited, maybe i got OK/UNREACHBALE status because of quality=200 in my sip.conf
15:53.08joker_89is esay to do that?
15:53.19[TK]D-Fender<joker_89> [[TK]D-Fender] which is the use of a GSM dongle for asterisk :S? <- to let your server talk directly to a GSM provider
15:53.30Milencoits pretty similar to using another channel in Asterisk really
15:53.48Milencofor Asterisk it shouldnt matter that much, but theres hardware involved and that always depends
15:53.50moe`if you check my sip.conf it has a [skype] block
15:54.02moe`and extensions.conf has a exten entry that refers to it
15:54.17Milencoindeed, so just try to call an number and see what happens
15:54.35Milencoas long as that exten => for skype rule is in extensions.conf, it should probably work
15:54.43[TK]D-Fender<[TK]D-Fender> moe`, Show. Us. The. Failed. Call.
15:54.52Milencoas in, i dont see any red flags just scrolling through your log
15:55.17Milencojust some minor things which shouldnt break it
15:55.27[TK]D-FenderMilenco, are yougetting them in PM?
15:55.36[TK]D-Fenderbecauxe I'm not seeing anything.......
15:55.38Milencofor example, type=friend, context=unixninja is already inherited from the friends-unixninja above
15:56.09MilencoNo incoming PM's here on IRC [TK]D-Fender, why do you ask?
15:56.30moe`what was the CLI for SIP debug again?
15:56.35Milencorasterisk -vv
15:56.39Milencoor more v's
15:56.42moe`god I'm sure I'm annoying the hell out of you guys
15:56.43Milencoor less :P
15:56.44[TK]D-FenderMilenco, because I asked for the debug 20 minutes ago and you seem to have gotten it
15:56.49Milenconah im bored anyway :P
15:57.02[TK]D-Fender<Milenco> rasterisk -vv <- nope
15:57.09[TK]D-FenderV != sip debug
15:57.12Milencosip show debug
15:57.20[TK]D-Fender"sip set debug on" <-
15:57.32Milencoyeah i know, but in case of moe` its unrelated to sip debug at this point
15:57.35[TK]D-Fender</inning>
15:57.37Milencoits just dialplan stuff
15:57.40[TK]D-Fenderis it?
15:57.44joker_89so, if i have a gsm modem with a SIM card, i can use that for mobile phone outgoing calls???
15:57.46Milenco'sip show debug peer bla' also ;)
15:57.47[TK]D-FenderWish I could see what I asked for...
15:58.01[TK]D-FenderMilenco, sip SET, not "sip SHOW.
15:58.04moe`yeah asterisk -rvvv gives some info, but not what [TK]D-Fender wants
15:58.08Milencohaha indeed :P
15:58.14moe`what's the CLI to enable debug again?
15:58.19[TK]D-Fendermoe`, You were vague on what the failure was
15:58.19Milencolittle bit wuzzy here :P
15:58.35[TK]D-Fendermoe`, at least show us the failed call here now.
15:58.37Milencomoe`, like [TK]D-Fender said above: sip set debug on
15:58.43moe`there we go
15:58.46moe`ok, hang on
15:58.48[TK]D-Fendermoe`, lets leave SIP out of it for now
15:58.55[TK]D-Fenderand see what basic verobse is showing
15:58.57Milencoindeed, just show what the dialplan does first :)
15:59.09Milencowhat [TK]D-Fender said
15:59.23moe`ok, so what exactly are you asking for then?
15:59.27moe`now I'm confused
15:59.30Milencojust open up rasterisk -vvv
15:59.32moe`situation normal
15:59.37Milencoand dial the number to dial over Skype
15:59.38[TK]D-Fendershow us the call from CLI without SIP debug for now
15:59.41Milencoand show us what the console shows
15:59.49moe`ok
15:59.51[TK]D-Fender~pb
15:59.51infobotA "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.
15:59.51moe`hang
15:59.52[TK]D-Fender^^^^
16:00.31joker_89where can i see the price of a gsm dongle for asterisk??
16:00.52Milenconot really sure, google for asterisk gsm dongle or something like that :P
16:01.17moe`https://unixninja.net/files/asterisk.messy
16:01.46Milencooke, so the dialplan setup is correct
16:02.09Milencoperhaps skype needs a +1 if you are dialling to the us
16:02.36Milencodo this instead and try again:
16:02.46Milencoexten => _XXXXX.,1,Dial(SIP/skype/+1${EXTEN})
16:02.52[TK]D-Fendernow we need the SIP debug
16:02.56[TK]D-Fender"sip set debug on"
16:03.11Milencoyeah im just guessing, sip set debug on would show a more accurate message
16:03.15[TK]D-FenderAnd don't bother piping * output to text lie that
16:03.28moe`I'll do the SIP debug as [TK]D-Fender has been asking for forever
16:03.29[TK]D-FenderYou get ANSI colouring crap which littlers your output with garbage
16:03.46[TK]D-Fendermoe`, now I can see we need it.  You are actually trying to call out a peer
16:04.28[TK]D-Fendermoe`, basic verbose would tell us if you screwed up normal dialplan processing to even get to this point.  You are calling Dial in a sane way at least so now it's down to the actual comms you are having with them
16:04.59Milencoindeed, Skype is receiving the call but refusing it
16:04.59moe`ok, cool, hang, I'll get you a "script" output of the call attempt here in a second
16:05.42moe`https://unixninja.net/files/asterisk.messy
16:05.46moe`refresh the page
16:05.49moe`it's the SIP debug
16:05.51Milencoin that other link i've send you this guy got it working with adding country code, like +1
16:06.22[TK]D-Fendermoe`, [0KReliably Transmitting (NAT) to 63.209.144.201:5060: <- first your provider is NOT behind NAT
16:06.27joker_89[Milenco] a usb gsm model will work?
16:06.29[TK]D-Fenderthey should be "nat=no" in your peer config
16:06.46Milencoyes joker_89, usb models can work like pci models
16:06.55moe`so sip.conf for [skype] block nat = no, ok
16:07.01[TK]D-FenderTo: <sip:15068517778@sip.skype.com> <- next they are VERY unlikly to like the FROM you are sending them.  Most providers expect you to pass a REAL looking phone number.
16:07.05Milencoas long as the system can interact with it; that is, drivers etc are isntalled
16:07.20[TK]D-Fenderor your username.
16:07.27moe`[TK]D-Fender - how would I do that, perchance?
16:07.37joker_89[Milenco] but how the VOIP call will work if the GSM card is not VOIP ?
16:07.39[TK]D-Fenderset "fromuser=YOURUSERNAME"
16:07.56[TK]D-Fenderjoker_89, because it is just GSM.
16:08.02moe`well I have that in the [skype] block in sip.conf
16:08.11[TK]D-Fenderjoker_89, the same as I can have a PCI card in my server and a dumb analog line plugged into it.
16:08.14moe`https://unixninja.net/files/sip.conf
16:08.17Milencoyou are mixing concepts joker_89, Asterisk != VoIP. Asterisk can handle all sorts of different phone types
16:08.23[TK]D-Fenderjoker_89, no voipo there.  it's dumb analog into PCI.
16:08.48[TK]D-Fenderjoker_89, So that US>GSM interface... does EXACTLY that
16:08.57[TK]D-FenderUSB
16:09.10moe`gets another beer
16:09.11[TK]D-Fendermoe`, evidently not.
16:09.16moe`not?
16:09.19[TK]D-Fendermoe`, Show us the config
16:09.25Milencohttps://unixninja.net/files/extensions.conf
16:09.31Milencohttps://unixninja.net/files/sip.conf
16:09.34moe`yeah
16:09.50[TK]D-Fender[skype](friends-unixninja) <- BAD
16:09.53[TK]D-Fenderstop using templates
16:09.54joker_89so, if I make a call using the GSM sim card dongle anyone more can make another external call using the same interface (sim card)
16:09.56joker_89?
16:10.05[TK]D-Fenderthat stuff you have in there does NOT relate to your skype entry
16:10.06Milencoreally, why? i use it all the time
16:10.22[TK]D-Fenderdoing it VERY wrong
16:10.25Milencoah, right, i dont use template for providers usually
16:10.41[TK]D-Fenderusername = 99051000268278
16:10.41[TK]D-Fenderfromuser = martin.unixninja
16:10.46[TK]D-Fenderthisis not what I said
16:11.00[TK]D-Fenderuse the USERNAME value for the FROMUSER
16:11.07[TK]D-Fenderusername = 99051000268278
16:11.13[TK]D-Fenderfromuser = 99051000268278
16:11.13moe`oh
16:11.21Milenconow i feel stupid
16:11.24Milenco:D]
16:11.25[TK]D-Fendernat = yes <- and NO
16:11.36[TK]D-Fender;exten => _XXXXXXXXXXX,1,Dial(SIP/skype/${EXTEN}) <- and this does not belong in sip.conf
16:11.52moe`yeah Milenco already mentioned that I believe
16:12.10[TK]D-Fendercould be.  I'm sure I skipped a few lines waiting for the bits I requested
16:12.19moe`HOLY SHIT GUYS
16:12.21moe`it's ringing
16:12.24Milencohaha
16:12.31moe`you guys are AWESOME
16:13.01[TK]D-FenderYou're welcome
16:13.01moe`that's goddamned awesome
16:13.03Milencowell, [TK]D-Fender mostly
16:13.06Milencoglad its working :)
16:13.22moe`holy crap too bad I can't DCC you each a case of beer
16:13.34Milencodont drink beer anyway
16:13.43Milenco*finishes rolling another joint*
16:14.07moe`ah, I hear that, just did it a few moments ago
16:14.20Milencoim mostly impressed with [TK]D-Fender ability to read that sip debug log so well
16:14.32moe`impressive indeed, he nailed it
16:14.38moe`within moments
16:14.50[TK]D-FenderThere's a reason I ask to see the call immediately
16:14.52[TK]D-FenderIt's all there.
16:15.12Milencowhat do you look for exactly in those debug logs?
16:15.38Milencoi always hate to look into it because all the debugs are coming on at the same time and i dont really know what to look for anyway
16:15.54[TK]D-FenderWho it's from, who it's to.  What IP's codecs, etc are offered.  Whether they are treated as being NAT'd, the number being dialed itself.
16:15.58[TK]D-FenderThe response order
16:16.02[TK]D-FenderALL of it
16:16.22[TK]D-Fender"that FROM: looks like crap"
16:16.39[TK]D-Fender"That number you're dialing is mangled and not normal, they won't like it"
16:16.40Milencoyeah im still looking at moe his debug log
16:16.41[TK]D-Fenderetc
16:19.52Milencohow could you tell the provider wasnt behind nat?
16:20.40Milencoi usually specify nat for my external providers because asterisk doesn't have its own external ip and goes through a firewall for 'internet' access
16:21.52Milencohttp://cdn.meme.am/instances/500x/57860290.jpg
16:23.45[TK]D-FenderMilenco, NO real provider ever is
16:24.10[TK]D-FenderMilenco, you have mixed up YOU being behind NAT... and THEM being behind NAT
16:24.18[TK]D-FenderYOU might be... THEY aren't
16:24.55Milencoaha
16:25.13Milencothanks, i'll fix that on monday
16:25.41Milencoand read up on the sip.conf documentation..
16:29.36Milencobut how does that interact with externalip= in sip.conf? VoIP-info says: "Indicates the IP address (alternatively you can enter a hostname) that will be used as the source IP address for all SIP messages when NAT is specified."
16:30.34[TK]D-Fenderin a peer definition it has no relationship
16:30.47[TK]D-Fenderthat affects the one under [general] <-
16:31.09[TK]D-Fender"nat=" is a parameter both for [general] AND for peers
16:31.16Milencoaha
16:31.20[TK]D-Fender[general] = YOU.
16:31.24Milencohow does it work for [general] then
16:31.25Milencooke
16:32.24Milencoso in my case, i should use nat=yes, localnet=10.0.0.0/255.0.0.0, externalip=x.x.x.x in my [general], but nat=no in my providers peer config?
16:32.41Milenco(assuming we use 10.0.0.0 for local stuff)
16:33.53Milencoit makes total sense btw now that you differentiate the both nat's but i have never thought of it like that
16:37.12*** join/#asterisk tzafrir (~tzafrir@bzq-179-40-172.cust.bezeqint.net)
16:37.39[TK]D-Fenderexternalip is not the proper name...
16:37.45[TK]D-Fenderthe rest seems sane
16:39.45*** join/#asterisk CeBe (~CeBe@port-92-200-233-239.dynamic.qsc.de)
16:39.54*** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de)
16:40.11Milencooke, thanks :)
16:40.48Milencoseems to be working fine now, but apparently thats more luck than wisdom
16:41.00moe`wow guys, this works awesome
16:41.07moe`I cannot thank you enough
16:41.16moe`so good
16:41.17Milenconp, are incoming calls working as well?
16:41.27moe`Milenco, I haven't gone there yet
16:41.44moe`I'll need to setup a skype in number
16:42.02moe`and in the near future, there will be several numbers, which should ring different extensions
16:42.21Milencoindeed, as for your extensions.conf, add a [public] context to handle the incoming call
16:42.33moe`ok, thanks for that tip
16:42.39moe`would not have thought of that
16:42.55Milencoand you possibly need to redirect it to a matching number in that [public] context, you can do so by adding a /123 in sip.conf after the register=> line
16:43.08Milencoand match that with a exten => 123 in [public]
16:43.09Milencofor example
16:43.42Milencocontext=public is your default context, so that is where the register => calls will come into
16:44.12Milencoanyway gl, im off for a shower
16:45.12[TK]D-Fender<Milenco> context=public is your default context, so that is where the register => calls will come into <- nope
16:45.20[TK]D-Fenderthis is not the basis of inbound calls.
16:46.26moe`[TK]D-Fender ++
16:46.29moe`you da man
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16:51.57[TK]D-FenderMilenco> and you possibly need to redirect it to a matching number in that [public] context, you can do so by adding a /123 in sip.conf after the register=> line <- this tells the place you are registering to to send to extension 123
16:52.05[TK]D-Fenderthey might not respect that request.
16:52.12[TK]D-FenderNext, as to where it actually goes....
16:52.24[TK]D-FenderWhen a call comes in it is matched against your peers.
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16:53.30[TK]D-Fenderfor "type=", peer matches by IP ADDRESS, user matches by the FROM: user they send, and friend acts like peer AND user combined
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16:54.28[TK]D-Fenderwhen the IP/user it is trying to match fits, an entry it then uses the auth rule you have set for it (secret + insecure settings).  If it PASSES then the call is accepts
16:55.05[TK]D-FenderNow if you have a "context=" specified in [general] and NOT in your entry, then [general]'s will apply, otherwise it takes what is in your peer
16:55.16[TK]D-Fenderand that is where the call is sent
16:55.38moe`gives [TK]D-Fender a beer
16:55.52moe`dude, the level of support you offer is awesome.
16:56.42moe`I'm still enjoying this newfound outbound call ability from my asterisk instance  :)
16:57.51moe`I would hope there's a paypal method to donate to the asterisk project?
16:58.20filethere is not
16:58.54moe`oh, not good.  there should be.
16:59.13moe`I'd fire off a bill to that
16:59.39filethe project doesn't quite operate like that
17:00.18moe`well I have very high recommedations for you guys, and asterisk
17:00.21fileas it's backed by a company we pay for a lot of things ourselves
17:00.53[TK]D-FenderAnd I'm regularly a masochist and sometimes a consultant
17:01.51filehelping others is always good, contributing documentation, etc
17:04.24moe`there is no way I could get this level of direct support from a big name like IBM or M$ or Oracle
17:04.48moe`absolutely no way
17:05.23moe`you guys took a good thing and made it even better, if that's even possible.
17:05.26nbjoergmoe`: sure there is
17:05.33nbjoergmoe`: whether you can afford it...
17:05.45moe`nbjoerg: indeed
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17:17.23Milencothink thats basically because its open source
17:17.32Milencoand therefor more community driven
17:17.50Milencoeven if its backed by Digium
17:21.05[TK]D-FenderIt's more Digium driven
17:21.13[TK]D-Fenderbut fed by the community
17:21.36[TK]D-FenderCommunity does most of the reporting, Digium does most of the doing
17:22.22Milencoim doing my dcap in september and while i felt pretty positive about doing it, now im getting more and more nervous getting these fact checks here :D
17:23.32filecommunity does reporting and features, we do more of the big picture things
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18:04.07moe`aw, suckage, skype does not offer an inbound number for the locations I want
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19:22.17tompawHello
19:23.10tompawIs there any way to put all non-admin members of a ConfBridge on/off hold without using AMI to query for members first?
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19:30.06[TK]D-FenderYou have to poll somehow yourself
19:34.50tompaw[TK]D-Fender: thanks.
19:35.33[TK]D-FenderYou know all those commands are completely documented
19:35.50[TK]D-FenderIf you donèt see a Èdo this to all non-adminsÈ... then it doesnèt exist
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19:39.22tompaw[TK]D-Fender: got it, I already keep track of all admins entering/leaving via AMI, so I'll have to extend it to regular users, too, and use it to hold/unhold them.
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20:33.32tompawErm... how do I put a selected user on hold anyway?
20:33.43tompawThere is no ConfBridge AMI action to do so.
20:34.01tompawI can mute/unmute them, but they will still be able to hear what's happenning in the conf room.
20:52.56tompawIs there even a way to put a selected PJSIP channel on hold via AMI?
21:00.28tompawSo it seems ConfBridge creators didn't think of something as simple as putting participants on temporary hold and 3rd party products are needed to do so :/
21:04.59tompawDoes this channel do bounties?
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21:10.19mjordantompaw: I wouldn't say it wasn't thought of. I would say you're doing things with ConfBridge that go beyond simple conferencing.
21:10.35mjordantompaw: You may want to look at BridgeWait, and ChannelRedirect/Redirect (AMI)
21:10.47mjordantompaw: or, just use ARI, which contains a /hold operation on the /channels resource
21:10.47tompawmjordan: funny, I'm looking at Redirect now :-)
21:10.55mjordantompaw: almost like we did _did_ think of things... :-P
21:11.16tompawmjordan: Do you really think this is out of regular business scope for a conferencing application?
21:11.25mjordantompaw: as for bounties - https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties
21:11.37mjordanI would say that most conferences I've used have mute, but not hold
21:12.00mjordangenerally, if you have the concept of leaders/followers, that is handled by marked/unmarked users
21:12.05mjordanand the "hold" happens when you have no marked users
21:12.11mjordanas opposed to a triggered action
21:12.35mjordanif you need something more complex than that, that's why we invented ARI. Because hard coding business logic to try to account for everyone's needs is silly.
21:12.49tompawI beg to differ - most of the times when you call big companies you end up being put on hold while another manager joins the conf, then they put you off hold, the original agent leaves and you continue your case with the manager.
21:12.53mjordanNow then, if someone wants to write a DTMF action that puts something on Hold, I think that's an interesting feature, and it is an open source project.
21:13.04mjordantompaw: well, then luckily Asterisk is open source, and we accept patches :-)
21:13.17mjordantompaw: but there are plenty of mechanisms that allow a more complex system to do what you want.
21:13.45tompawAMI REdirect being one of them?
21:14.00mjordanRedirect removes a channel from its existing application and puts it into another location in the dialplan.
21:14.01tompawlike - redirect them to a music on hold extension and then pull them back into confbridge?
21:14.09mjordanIf you want a temporary holding area, then use BridgeWait in that extension
21:14.17mjordanAnd when you want to put them back in the Conference, you'd Redirect them back in.
21:14.34tompawAlmost like if you did did think of it, indeed :P
21:14.57mjordanWell, as I said, if I were you, I'd just use ARI, as making a conference in ARI is relatively trivial, and then you have full control over the resources.
21:15.30mjordanheads off to Target
21:15.39tompawI'd look into it, but few weeks ago when I asked about AMI/ARI to do what I need (i.e. track users that join/leave conferences), most people suggested the AMI approach.
21:15.44tompawmjordan: thanks
21:17.31filedepends on why you are using ConfBridge, if you have to use it - then AMI is it but if you are using ConfBridge as a handy way to connect things together then ARI provides that functionality and allows you far greater control (but more responsibility for implementing features)
21:17.41WIMPyThat sounds like just a normal 3PTY action. Used to be really simple.
21:27.08mjordanWIMPy: I'm not sure what you're referring to, since nothing's been removed, and Asterisk hasn't had the ability to do an ad-hoc multi-party bridge until 13.
21:27.31mjordannow really does go off to Target
21:27.36WIMPyNot from Asterisk. From Telephony before Asterisk.
21:28.17WIMPytompaw is rigt this is a standard scenario and without Asterisk it's just a press of a button.
21:28.33WIMPyThose were the days...
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21:50.14joker_89hi
21:50.26joker_89[[TK]D-Fender] and Milenco are both alive?lol
21:52.34joker_89if I connect a gsm interface to asterisk and configure it to use it for outgoing calls to mobile phone numbers. Can I make to use the DID number for the call receptor?
21:52.43joker_89instead of the mobile phone SIM number
21:53.24WIMPyno
21:53.45joker_89so, does they see the mobile phone number, right?
21:54.36WIMPysure
21:55.30joker_89then, how can I use some outgoing calls VOIP provider that has a unlimited calls for mobile phone numbers in my country

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