IRC log for #asterisk on 20140214

00:11.55*** join/#asterisk Pullphinger (~Pullphing@c-24-13-69-42.hsd1.il.comcast.net)
00:20.13*** join/#asterisk jetlag (~jetlag@pool-71-168-240-196.cmdnnj.east.verizon.net)
00:20.48DevWork_Haven't really figured out setting caller id. I got my pbx taking calls and routing them out. On my android if I set the callerid to "Asterisk" <1${CALLERID(num)}> I see just ${CALLERID(num)} no one on the phone. My idea was, I would see some number on a normal call, and 1NUMBER on a call from the pbx routed to my cell.
00:21.16DevWork_So I can set the callerid to a number, any number, but its difficult to distinguish it from an actual call from that number
00:21.34DevWork_And I definately havent got anything regarding the name portion working.
00:22.32[TK]D-FenderYou almost never get to set a name on outgoing
00:22.41DevWork_Ok so number is fine.
00:22.58*** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk)
00:23.26DevWork_I read a document that mentioned basically using 1 followed by the number from the PBX, and no one for a normal number ( since normal numbers never start with a 1 )
00:23.33DevWork_And that sounded cool.
00:23.43DevWork_I just didn't know if that worked or other similar ways to do that.
00:25.22[TK]D-Fenderyou need to keep where you are putting these out to in mind
00:25.36DevWork_Yea thats fine.
00:25.46[TK]D-Fenderchanging names and prefixing the number, etc works fine if you're calling a SIP device... the PSTN's rules are another matter entirely
00:26.13DevWork_So what can you do with PSTN numbers?
00:26.56[TK]D-Fendergenerally just the number
00:27.28[TK]D-Fenderand in many cases a provider may prevent you from coming up with "just anything" and restrict you to #'s that match DID's you pay them for
00:27.35DevWork_So theres no way to squeeze any information in there, besides manipulating the 10 digit number
00:27.40[TK]D-FenderSome you can tell to allow certain specific exceptions.
00:27.56[TK]D-FenderYou're calling the outside world... you don't have a lot fo control
00:28.09DevWork_Yea, I get that.
00:28.21DevWork_But it is letting me set any 10 digit caller id number I want.
00:28.52[TK]D-Fenderlucky you....
00:29.24DevWork_I wonder what I can do with the first digit tho.
00:29.35DevWork_Maybe 0 if its from the system, and normal number otherwise.
00:30.06DevWork_does syntax like ${CALLERID(num):1} work?
00:31.53[TK]D-Fenderthat is A syntax....
00:32.03[TK]D-Fenderfor a substring....
00:33.31DevWork_Sweet. I can add a 0 to the full number and it will display it all.
00:34.37[TK]D-FenderTmay or may not work....
00:34.46DevWork_It did work, I just did it.
00:35.02[TK]D-Fenderand you saw it the way you wanted on inbound?
00:35.20DevWork_When my cell phone range, it had 05555551234
00:35.24DevWork_rang*
00:35.52[TK]D-Fenderalrighty...
00:36.01DevWork_So thats exactly what I wanted to do, and it is working.
00:38.34DevWork_Is it a common issue that is unavoidable that when I dial my DID, that it takes a little bit to hear the ring?
00:38.49DevWork_like I would have to switch provider or something, type of unavoidable.
00:39.19[TK]D-FenderYou're likely to get a small delay from most
00:40.00[TK]D-Fender1-2 sec tops
00:41.12DevWork_ill time it i want to look at my firewall though when i do
00:42.32[TK]D-Fenderthat sannot realistically add enough to be perceptible
00:42.35[TK]D-Fendercannot*
00:43.00DevWork_if my firewall is blocking it until my end pings back out to the ISTP, it could be.
00:44.03DevWork_Its actually not a problem though, where I work our existing pbx uses voip to go out. When I do that, it takes 8-9 seconds to ring. When I call my DID from my cellphone however, i hear the recording almost immediately.
00:46.08*** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183)
00:46.20DevWork_damn ive never realized how slow our phones are outbound at work.
00:52.19[TK]D-FenderWhat "existing pbx"?
00:52.43DevWork_meridian nortel
00:52.50DevWork_1980's junk.
00:53.51DevWork_Is there a way to see what the dialplan parser is choking on?
00:57.12[TK]D-Fenderyou should be a lot more detailed about the scenario...
00:57.23[TK]D-Fenderyou just said existing PBX over VoIP.
00:57.36[TK]D-FenderOld Norstar didn';t speak voi and you haven't been clear about how it gets there
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04:40.58j35ureHey everyone! Had a quick question tonight, I hope, heh. Whats the state of faxing ( incoming and or outgoing ) in asterix? Is it doable on VoIP reliably? Can I do it with a dialtone on a normal pstn from the wall via the telco into my asterisk server reliably?
04:43.03j35ureWhat it comes down to is I have to send faxes, and it is just not worth it to me to worry about if the other side got it or not.
04:45.49[TK]D-FenderOver VoIP requires T.38.  It's usually pretty good then
04:46.07lnb_Why is it the case involving SIP URI's where a remote calls sip/sales@pbx.domain.tld where sales is an alias for extension 300 and it just so happens the callers ext is 300 and asterisk looks at it as if the remote caller is the local extension and wants it to AUTH?
04:46.13lnb_[TK]D-Fender: IT SUCKS
04:46.30lnb_class repeat: VoIP is NOT FOIP
04:46.38*** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183)
04:47.20[TK]D-FenderDepends on a certain point of view.
04:47.44lnb_j35ure: try a 10 or more page fax outbound. Then -> Class Repeat after me: Voip is NOT FOIP
04:47.51[TK]D-Fenderj35ure: as for your PSTN question... depends exactly what you use to get to that jack.  Then there is the questio about what is actually doing the sending.
04:48.37lnb_[TK]D-Fender: try sending out a 15 page fax and let me know how it goes.
04:49.42[TK]D-FenderForgive me if I take you as the example of every way NOT to manage a system....
04:50.00lnb_the other problem is, faxing even with proper t.38, goes AT BEST 14400. where as analogue goes out at 37k
04:50.52[TK]D-FenderDepends how you define that as a "problem".
04:51.14lnb_problem=takes too long, IF you can get past page 5
04:51.30lnb_4=< is ok
04:51.52[TK]D-FenderToo long?  For you maybe.  Maybe others don't care.
04:51.58lnb_ummm
04:52.12lnb_do the math for 100 page fax
04:52.35lnb_my clients send out 100+ pages for every one of their clients
04:52.40lnb_medical
04:52.43[TK]D-FenderYou do the math and consider how many people have to do that.
04:52.53lnb_i dont wana do the math
04:52.58lnb_you do it
04:53.18lnb_i just know it doesnt work well period
04:53.27lnb_and i am not alone on this either
04:53.55lnb_hey, i only send out 3-5 pages, and most times its just fine
04:54.19lnb_except to canadian gov't
04:54.28lnb_forget sending there
04:54.36j35ureSo you need T.38. Does everyone have T.38? Is that a normal thing now? Or are people going to not be getting faxes if I send them with that stuff.
04:54.38[TK]D-FenderYou have that guy who faxes out 100+ pages at a time.  That is puny % of the worlds needs.
04:54.42lnb_just do yourself a favor and have a good German beer
04:55.00j35ureWe reach our fax machines memory limit weekly ( 50 pages ).
04:55.09lnb_doesnt matter [TK]D-Fender its a fact
04:55.15[TK]D-FenderA useless fact
04:55.24lnb_and the medical and insurance industries are HUGE
04:55.27lnb_in your mind
04:55.50[TK]D-Fenderdo you think Joe Blow just walked in here with YOUR guys needs?
04:56.04[TK]D-FenderYou're framing it like EVERYONE has the same needs
04:56.06lnb_what did i say above [TK]D-Fender
04:56.18j35ureSo faxes out, you need a dedicated machine on a regular phone jack with a dialtone it sounds like?
04:56.23lnb_>5 pages is OK
04:56.28lnb_oops
04:56.29[TK]D-Fender[23:55]lnb_and the medical and insurance industries are HUGE <- and how many people come in here in THOSE categories?
04:56.34lnb_< 5pages is OK
04:56.55lnb_do YOU know for sure who comes in here?
04:57.12lnb_i sure don't
04:57.19[TK]D-FenderAfter every chump sized install.. yeah, we've got a pretty good concept of scale
04:57.38j35ureSo how can I get reliable faxes out. Can I get Verizon to drop in an extra analog line and use that to send faxes out non T.38 so everyone can receive them?
04:57.40lnb_congratulations. to what address do I send the prize?
04:57.52[TK]D-Fenderj35ure: what is this about a machine's memory?
04:58.15j35ureMy fax machine can't do > 50 pages at once. We have to chop up those into multiple runs.
04:58.17lnb_j35ure: can i ask you, how many pages per fax do you need to send?
04:58.42j35ure1 - 200. The 200+ we currently do in sets of < 50.
04:58.53lnb_200 page faxes?
04:58.57[TK]D-Fenderj35ure: What are you sending?  how big would an individual transmission be if not limited in some other way?
04:59.14lnb_j35ure: 200 page faxes?
04:59.29j35ureWhat are we sending? Discovery documents. Contracts. Bids. Estimates. I'm not sure why it matters what it is.
04:59.41lnb_i did not ask the content
04:59.43lnb_j35ure: 200 page faxes?
04:59.49[TK]D-Fenderj35ure: What kind of overall volume?  Are you looking to basically plug in a machine you already have?
05:00.04j35ureLike I said, we break them into < 50 pages a fax, because the machines memory will quit more than that.
05:00.34lnb_j35ure: if you're planning on sending > 5 pages, USE ANALOGUE
05:00.41j35ure[TK]D-Fender, I'd like to scan them first, and then send them out via the PC. That is the dream. If we had to, we could get a fax machine and have it talk to asterisk.
05:01.05j35urelnb_, Analog as in not VoIP? Ok that is fine.
05:01.18[TK]D-Fenderj35ure: This is worth a dedicated line for
05:01.20lnb_j35ure: yes, your local telco
05:01.37lnb_j35ure: or join the 'bald' club
05:01.49[TK]D-Fenderj35ure: At which point it's up to you and some fax software... which leaves Asterisk completly out of this
05:02.25j35ureWell, the building I want to do this in, has 200 desks on 35,000 square feet. Ideally I'd like to let everyone send a fax from their PC.
05:02.39j35ureSo send it to asterisk and have it send it out from the server cabinet where the verizon line is.
05:02.42lnb_j35ure: thats not the issue
05:02.50[TK]D-Fenderj35ure: So set up a fax server for them then.
05:03.09j35ureYea that is why I'm here.
05:03.12lnb_j35ure: i can set up 500 person office to fax from the pc, but not with VoIP (asterisk/freeswitch)
05:03.24[TK]D-Fenderj35ure: there is no "send it to Asterisk"...
05:03.40lnb_[TK]D-Fender: thats a bit wrong
05:03.41j35ureI was going to use like an analog card from asterisk. I was told from an asterisk consultant that it did something like this.
05:03.42[TK]D-Fenderj35ure: Forget Asterisk, it is not part of your fax sharing solution.
05:03.56lnb_[TK]D-Fender: you can use schmoozecom fax pro
05:04.13lnb_but not for what j35ure is doing
05:04.13[TK]D-FenderI've seen it.  No thanks.
05:04.28lnb_i set it up in an office, they swear by it
05:04.40lnb_but they send up to 5 pages/fax
05:04.42lnb_no more
05:04.42[TK]D-Fender"That's nice"
05:04.47[TK]D-FenderPeople put up with a lot of stuff.
05:04.49lnb_yes, indeed it is
05:04.55[TK]D-FenderPeople don't know what "better" looks like
05:04.55lnb_put up?
05:05.03lnb_dude it WORKS
05:05.12lnb_for what they need
05:05.12j35ureSo whats the asterisk of the fax server world?
05:05.15[TK]D-FenderI've seen it, and I've seen far better
05:05.24lnb_name it
05:05.47[TK]D-Fenderj35ure: Lots of commercial stuff out there.  If you want to roll your own : Hylafax
05:06.33lnb_elastix uses hylafax
05:07.08lnb_but you must understand, you are sending at 14.4 whereas analogue is 33, 34..
05:07.16[TK]D-Fendersays who?
05:07.31lnb_G-D
05:07.47[TK]D-FenderYou are making assumptions as to what it goes over.
05:08.19j35ureSo do I have to run an analog line per fax line with something like hylafax? What hardware do you install, it says fax modems, I'm picturing the 90's and 56ks again here.
05:08.23lnb_[TK]D-Fender: prove to me, or as you say, show me the call of a fax using VoIP at 30k
05:08.48[TK]D-FenderI'm not talking about FoIP.  I haven't been for a long time
05:08.49lnb_j35ure: if you want to keep your sanity and maybe more, yes
05:08.54[TK]D-FenderWake up and follow the conversation.
05:09.08j35urelnb, Vague much? What do you mean by that.
05:09.40[TK]D-Fender[00:08]j35ureSo do I have to run an analog line per fax line with something like hylafax? What hardware do you install, it says fax modems, I'm picturing the 90's and 56ks again here. <- yup
05:09.54lnb_j35ure: i dont know what your position is with the company you plan on doing this with, but if you want to prove failure, use VoIP for those faxes, see where it gets you
05:10.11j35urelnb_, I know FoIP is failure man.
05:11.05j35ureI was under the illusion that something had changed in the faxing world, that I didn't need 5 analog lines from my telco. Now I am back to requiring monthly costs per lines again and that just sucks.
05:11.41j35ureThey use T1 fax boards.
05:11.46j35ureNow that we can do.
05:12.06j35ureAnything that works with asterisk as well?
05:12.24j35ureI can call my telco and get a PRI via T1 tommorow.
05:12.31[TK]D-FenderYou keep through "asterisk" in there.  Why?  What magical role are you expecting?
05:12.55j35ureUhh well, asterisk can uses some of the 23 channels on the T1, and the fax can use its own.
05:13.18[TK]D-FenderYou're asking separate from your faxing questions?
05:13.30lnb_j35ure: you can use freepbx in the office and connect it with digium board to your t1
05:13.48lnb_but i would not do it that wa
05:13.50lnb_way
05:13.59j35ureWhat would you do then lnb_?
05:14.25[TK]D-Fendersits back and grabs some popcorn....
05:14.39lnb_i would get analogue line for fax, and i would use asterisk/freepbx for voice calls, extensions, queues, voice mail, disa, and a ton more
05:14.54j35ureHere is the thing. We need a PRI. If we can use what we already have, and deliver faxes over it as well, that is the ideal situation. Getting 5+ phone lines just for faxes from the telephone company is a lot of extra money per month that seems super unnecessary.
05:15.04lnb_why do you need a pri
05:15.25lnb_you mean the whole 23 channels?
05:15.28j35ureBecause it lets me get 23 channels from my telco. I already need 20. I have 3 left over, why not use them for hylafax?
05:15.40lnb_i dont get it
05:15.45j35ureWhat do you not get?
05:15.57lnb_why are you using pots instead of voip?
05:16.05lnb_for your voice calls
05:16.13j35ureBecause to me, it is irrelevant if I use the PRI or VoIP.
05:16.22lnb_really
05:16.25lnb_hmm
05:16.30lnb_what about costs
05:16.36j35ure200$/m for the PRI.
05:16.48lnb_ok
05:16.55lnb_how many fax lines do you need
05:17.08j35ureIf its over PRI, I could get away with just using one channel.
05:17.18lnb_ok order 1 line
05:17.20j35ureIf it isn't, then I need minimum 5 analog lines.
05:17.32lnb_huh?
05:17.48lnb_you said you need 1 channel for fax
05:17.53j35ureI have 5 local + 1 toll free DID, all for faxes.
05:18.00lnb_so order 1 line for FAX
05:18.18lnb_what is your internet capability?
05:18.27lnb_in terms of bandwidth?
05:18.35j35ure20megabytes symmetric.
05:18.39lnb_ok
05:18.48lnb_how many concurrent calls?
05:18.51lnb_at most
05:19.24j35ureWe do 20 concurrent calls max in the past 360 days.
05:19.30lnb_ok
05:19.35lnb_your fine
05:19.44j35ureFine with what?
05:19.44lnb_order 1 or 2 lines for FAX
05:19.51lnb_fine with EVERYTHING
05:19.57lnb_order 1 or 2 lines for FAX
05:20.14lnb_and setup an asterisk/freepbx box in the office
05:20.20j35ureLike normal phone line dial tone rings?
05:20.21lnb_and save money
05:20.29lnb_for?
05:20.42lnb_<j35ure> Like normal phone line dial tone rings? for?
05:21.09j35ureFFS man. I have 6 phone numbers, they all receive faxes.
05:21.20j35ureHow does 1/2 analog lines from the telco help me.
05:21.22lnb_then you need 6 lines
05:21.30lnb_so order 6 lines from telco
05:21.35*** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183)
05:21.37j35ureYea and the monthly eats you alive.
05:21.50lnb_how can that be
05:21.59j35ureYou price a ringy ding analog line before?
05:22.00lnb_i have customers like that
05:22.06lnb_yes
05:22.07j35ureThey are 5$/month minimum.
05:22.09lnb_30.00/month
05:22.13lnb_thats all?
05:22.16lnb_wow
05:22.17j35ureOk that's like 20x worse then.
05:22.24j35ureSo I take my existing PRI.
05:22.25j35ureAnd one channel
05:22.27lnb_welcome to canada
05:22.29j35ureand I pay 0 more a month.
05:23.19lnb_so you are talking $30.00/month for 6 lines
05:23.25lnb_and you are whining?
05:23.47j35ureI said minimum.
05:23.52j35ureOurs are 15.
05:23.52lnb_wtf
05:24.02lnb_that is gurnicht
05:24.06lnb_that is NOTHING
05:24.09lnb_a joke
05:24.13j35ureSo that is 75$ a month. That is 900$ a year.
05:24.31lnb_to what address do i send the pink tissues?
05:24.48ChannelZwhat counties are you in?
05:24.56lnb_so get the pri and be done with it
05:25.15lnb_ChannelZ: right now just in one
05:26.09ChannelZI meant countries as you and j35ure talking about $5 analog lines (unless I misunderstood, havent read back all the way)
05:26.19lnb_i am in canada
05:26.23lnb_he did not say
05:26.38j35ureWest side, canada's neighbor.
05:26.51lnb_that would be the ocean
05:27.01lnb_the pacific
05:27.13j35ureI'm on the west side OF canada's neighbor. But you knew that.
05:27.16lnb_the controlling factor of temperatures
05:27.37lnb_j35ure: that makes no sense
05:27.50j35ureIt makes perfect sense.
05:27.56lnb_on either side of canada are oceans
05:28.04lnb_you might mean SOUTH
05:28.08ChannelZmy monthly phone bill for a single POTS line was more than $30/mo
05:28.16j35ureWho is canada's neighbor lnb_ ?
05:28.24lnb_Arabia?
05:28.29j35ureI'm on the west side of arabia.
05:28.45lnb_that would put you in Jordan
05:28.49lnb_how is the queen?
05:28.54lnb_she still hot?
05:29.05j35ureOfc.
05:30.44lnb_j35ure: u said you pay $5/line
05:31.01j35ureI said they are 5$/month minimum after I asked you have you priced them out.
05:31.06lnb_all i am saying is, take advantage of Asterisk/FreePBX for all your phone line
05:31.08lnb_lines
05:31.16j35ureI know where analog lines are 5$ a month, and we can't get them at my current local.
05:31.17lnb_use the telco for fax
05:31.24lnb_and you have a rock solid plan
05:32.33lnb_Martin says Hacker-Pschorr is 'delicious' and I say he is nuts. Erdinger is so much better
05:39.39lnbMein sehr lieber Freund Martin, es gibt keinen Vergleich zu Erdinger Weizen-Bier vom Hacker-pschorr
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05:47.56wacomcitogood morning on friday!!!! :)
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09:31.56mirela666does anyone know syntax for GotoIfTime, do i specify days with coma
09:32.11mirela666for example: ifTime(16:30-18:00|mon,tue,thu|*|*)
09:32.29mirela666well ael ifTime
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09:33.44kaldemarmirela666: core show application ...
09:42.23mirela666i did take a look but still don't undestand how to specify mon, tue, thu
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09:47.34kaldemarmirela666: you may need two gotoifs in that case. one with mon-tue and another for thu.
09:47.58mirela666yes, that is what i thought is needed, thanks
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09:48.15mirela666otherwise I get ael compilation error
09:48.24mirela666compliling*
09:48.27kaldemarmirela666: actually, you should be able to use "mon-tue&thu"
09:48.36mirela666aha
09:49.28kaldemartimes, weekdays, mdays and months are all lists of ranges, ampersand as the delimiter.
09:50.06mirela666damn not compiling
09:50.47mirela666<PROTECTED>
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09:52.11kaldemarare you using some old version?
09:52.32mirela666ast 1.6.2 :| yeah
09:54.01kaldemarmaybe you're stuck with two GotoIfTimes then.
09:54.03mirela666I'll use 2 else if-s then :) thx for help :0
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10:25.00IamTryingIs Asterisk works with RFC5766 turn server? I have problems with FreeSWITCH its not working in some NAT location
10:25.41IamTryingDid anyone made it work with Asterisk yet ? To cover the firewall/behind nat problems? FreeSWITCH is failing over and over.
10:26.14WIMPyWe usually configure nat=yes and nothing on the client side.
10:26.59IamTryingWIMPy, do you have a demo page? i can test? cause the PC and network i am testing nothing works (FreeSWITCH, Yate, WebRTC none of them working except Bria, PortGO softphone)
10:27.23WIMPydemo page?
10:28.29IamTryingWIMPy, Like this a demo page which is with Asterisk configured? (like here FreeSWITCH has which does not work https://webrtc.freeswitch.org/webrtc/portal.html )
10:29.35IamTryinghttps://webrtc.freeswitch.org/webrtc/portal.html - it does not work from the PC and network. But when i use same PC same network with Bria and PortGO softphone same FreeSWITCH works. I wanted to test if Asterisk failing or working so any demo page?
10:29.54WIMPyI have no idea.
10:30.25WIMPySIP is bad enough. I definitely don't want to know anythign about WebRTC.
10:30.54WIMPyTry to ask again later when the US guys are at work.
10:31.05IamTryinghttps://webrtc.freeswitch.org/webrtc/portal.html  - Nobody in Asterisk community made a sample demo page like this???
10:31.13IamTryingOK
10:31.45IamTryingWanted to compare Asterisk and FreeSWITCH, my guess is that FreeSWITCH has BUG which causing that failure.
10:32.03IamTryingBut me no Asterisk expert. So just wanted to use some existing demo page to quick debug and test.
10:32.32WIMPyOr the client side router is messing things up.
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10:41.26IamTryingWIMPy, client side router is messing things up since 30 years for SIP. But today many has fixed it even in messy router it works.
10:41.38IamTryingIn the same PC like i said others working except FreeSWITCH
10:43.41WIMPyWe know that NAT is evil, but I'm talking about ALGs. The mess up the messing up.
10:45.32IamTryingWIMPy, Consider that PC, Network is messed up broken + NAT EVIL+++. But still on that same broken PC, network Bria soft phone, PortGO working and WebRTC non FreeSWITCH based implementation working. Only FreeSWITCH demo page does not work.
10:45.55IamTryingDoes it not explain that FreeSWITCH is having a bug? So i wanted to compare if Asterisk demo page is doing same or not.
10:46.55WIMPyThat doesn't have to be a bug. In some other messed up situation it might get it right and the others fail.
10:47.34WIMPyThere are many ways to handle such situatuins. That doesn't make one that fails in one situation a bug.
10:47.43WIMPyBut it could be a bug, off course.
10:49.27sekilanyone doing asterisk and skinny?
10:49.51IamTryingWIMPy, same FreeSWITCH  official demo server is working with PortGo, Bria, same FreeSWITCH official demo webRTC link is not working at all
10:50.01WIMPysekil: There are multiple channels for SCCP, so someone must have tried it.
10:50.32IamTryingWIMPy, same demo page webRTC not using FreeSWITCH but using other Switch working from same broken PC, same broken network.
10:52.17IamTryingi think FreeSWITCH stopped fixing the ws:// and wss:// issues. SDP is not handling correctly. Wanted to test this same thing with Asterisk. Nobody made a demo page with * ?
10:58.04IamTryingCan Asterisk handle live concurrent calls of 300 session, over webRTC (Opus audio, VP8 video) ? e.g: if i use Asterisk and not use FreeSWITCH just because its unstable
11:02.04IamTryinghttps://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support - Here the Paragraph pjproject is it totally correct information? without pjproject no audio over webRTC? (what a crap is that mean). RFC5766 works so perfect what is wrong with that, instead of need to use pjproject?
11:02.43WIMPy???
11:03.03IamTryingWIMPy, it says:
11:03.05IamTryingAsterisk 11 comes with an embedded pjproject. When building Asterisk 11, to get ICE support you'll need the UUID development library (uuid-dev for Debian, libuuid-devel for CentOS) library. If you don't have ICE support, then you'll likely run into audio issues in several scenarios, specifically when attempting to traverse NAT, as WebRTC uses ICE,STUN,TURN to do this.
11:03.14IamTryingStarting with Asterisk 12 you need to have pjproject libraries installed, otherwise you most likely won't have audio in your WebRTC calls and no warning whatsoever!
11:03.49IamTryingWhy do i must use PjProject? Why can i not use native RFC5766-turn-server which works perfect.
11:04.10WIMPyWell, 12 should be viewed as experimental, I guess. But not giving a warning isn't nice.
11:04.39WIMPyAsterisk 12 uses pjproject for the new sip channel.
11:05.26IamTryingWIMPy, but they could use RFC5766-turn-server no? It works great with Google!!
11:05.58WIMPyWhat do the apples have to do with the oranges?
11:06.57IamTryingOk - you lost me WIMPy, i understand, we have to wait next 20 years to have SIP works 100% in any network like Skype.
11:07.27WIMPyNI don'tthin we will have SIP for that long.
11:07.42WIMPyI don't think we will have SIP for that long.
11:08.27IamTryinggod really bless SIP. its being completely screwed now still where Skype just kick those excuses of "sorry your router issue sorry firewall maybe!!"
11:09.15WIMPyWell, use Skype if you like it.
11:09.42mirela666hehe creators of Skype created the SIP :P
11:09.55WIMPyI'd go for IAX. Possibly the best thing from Digium. Unfortunatly they seem to think differently.
11:11.06WIMPyDUNDi is also a great thing. It's just missing a working implementation.
11:11.14IamTryingWIMPy, mirela666, still today the largest Asterisk, FreeSWITCH, Yate. Has not a single workout of Skype. Its like Shame in front of Skype. Look at Google, Asterisk should have done this before google came.
11:12.15WIMPyWell, Skype decided they want to be on their own.
11:12.52mirela666yeah t is Mcrosoft now, what to expect
11:12.54IamTryingOK - but today we have WebRTC. Why not jump on it ?
11:13.05IamTryingAsterisk still guessing PjProject or WebRTC?
11:13.15WIMPyI think YOU just gave the reason for that.
11:13.21IamTryingits being ages/decades SIP deserves a kick in network/firewall
11:14.01WIMPyThe only kick SIP deserves is to nirvana.
11:14.23WIMPyBut that seems to come from within now.
11:14.42IamTryinghttps://webrtc.freeswitch.org/webrtc/portal.html - NO - just make this work for any network/firewall as simple as that. Asterisk or FS no matter as long as it works like Skype in Chrome.
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11:15.46WIMPySkype isn't always P2P. It handles difficult situation by letting a 3rd party handle the traffic. That requires someone to pay for it.
11:16.13WIMPyAnd that usually works with SIP as well.
11:17.04IamTryingWIMPy, Everyone is ready to do that pay. Since 30 years ago. But there is no best workout
11:18.13IamTryingAsterisk still today do not have webRTC 100% firewall/network break implementation. ITs like next 20 year we have to wait for it, in total 50 years plan to do same thing what Skype did. Feel like shame on SIP
11:18.41WIMPyI didn't have any issues so far when directing the traffic through a central server.
11:19.10WIMPyPossible. So either you wait or you use something else, no?
11:22.42IamTryinghttp://zingaya.com/ WIMPy, someone already implemented with Asterisk something and that works
11:23.41IamTryinghttp://zingaya.com/  works anywhere but https://webrtc.freeswitch.org/webrtc/portal.html does not work  why?
11:25.11WIMPyWireshark is your friend.
11:26.26IamTryingWIMPy, Asterisk they used as far as i know. But FreeSWITCH when used its not working for me. TCPdump shows that SDP maybe the real cause but i am still not sure sometimes the ports are matching but no media
11:27.29mirela666<PROTECTED>
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11:27.37kaldemarif you know they use asterisk, there's your answer.
11:28.04ermali82hi there
11:28.30IamTryingmirela666, YES sipml5 to IVR of the Asterisk to make sure it works but its not working when i tested this: https://webrtc.freeswitch.org/webrtc/portal.html
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11:28.54ermali82does anyone know why asterisk sometime sends 2 consecutive invites to a peer (no nat involved)
11:29.54mirela666try with http://sipml5.org/call.htm?svn=203
11:30.10ermali82after some seconds the remote peer drops the call with "maximum retries exceeded on transmission..."
11:33.26mirela666ermali82: souds like asterisk is doing the re-invite
11:33.38kaldemarermali82: that's asterisk dropping the call, not the remote.
11:34.11mirela666ermali82: try with canreinvite=no anddirectmedia=no for that peer
11:34.51IamTryingmirela666, http://sipml5.org/call.htm?svn=203 - i apply the settings and it say: failed to connect to the server
11:35.27ermali82@kaldemar take a look to the pastebin http://pastebin.com/FCEeLnLk
11:35.37mirela666IamTrying: what * version are you using for test?
11:36.38ermali82in fact it is a trunk between 2 asterisks
11:36.42IamTryingmirela666, git clone git://git.asterisk.org/dahdi/linux dahdi-linux
11:37.05WIMPyermali82: Why don't you use IAX? That's what it's there for.
11:37.47mirela666IamTrying: asterisk -rx "core show version"
11:37.50ermali82becouse i send some information in the SIP headers
11:38.32WIMPyThat's possible in an even cleaner way via IAX.
11:39.28ermali82<WIMPy> How to send custom headers via IAX ?
11:39.35WIMPyIAXVAR()
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11:40.28ermali82<WIMPy> and how do i read on the remote end?
11:40.37kaldemarermali82: a sip debug would show what's really going on.
11:40.51WIMPyIAXVAR() again
11:42.21ermali82WIMPy thank you
11:42.24ermali82i will try
11:42.56ermali82kaldemar did you see the pastebin
11:49.50kaldemarermali82: yes. a sip debug with verbosity will tell more.
11:50.26ermali82it is a havy load asterisk so it is dificult to do a sip debug
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11:55.08ermali82kaldemar here the dump from the remote server http://pastebin.com/mvZEC6ng
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12:46.43wonderworldermali82: which tool do you use to create this presentation of the SIP session? looks very useful...
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13:24.15ermali82wonderworld wireshark export of tcpdump capture
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14:22.31anonymouz666friday... time to... super street fighter IV arcade edition!
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14:24.31windbackI have just installed dahdi 2.9 and I can't see anymore the file /etc/udev/rules.d/dahdi.rules. Does somebody know how can read write /dev/dahdi as non-root user?
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14:25.28ChainsawI'm not aware of a 2.9 being out.
14:25.43ChainsawAh, 2.9.0.1
14:25.57ChainsawWhoever keeps this topic up to date needs more caffeine :)
14:26.00windbackyes 2.9.0.1
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14:29.15*** topic/#asterisk by leifmadsen -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op
14:29.21leifmadsenit should be the person releasing the tarball
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14:31.20Chainsawleifmadsen: Well thank you all the same :)
14:33.26windbackleifmadsen, Do you know how to run asterisk as non-root with access to dahdi ?
14:33.33leifmadsennope
14:33.39leifmadsenmaybe it's documented ;)
14:33.44leifmadsenI don't remember off hand
14:34.09windbackleifmadsen, thanks.. I will continue investigating
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14:39.25Kattywhere's my gerunds at?!
14:39.45Kattypeople be stealing my infinitives and participles
14:40.27[TK]D-FenderPlaying in the street ;)
14:40.56[TK]D-FenderI proposed "Split Infinitives" as a band name once...
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14:41.39Kattythat'd make a lovely band name.
14:42.23[TK]D-FenderSpeaking of which.... launch day today!  woooo
14:42.42Katty[TK]D-Fender seems to /really/ like it, too
14:42.58Kattywhat are you launching?
14:43.04[TK]D-FenderMy new band... 1st show
14:43.12Kattydo you have the butterflies?
14:43.47[TK]D-Fenderthe excited kind, not nervous.
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14:44.15[TK]D-FenderI've been in plenty and I channel while playing...
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14:47.15[TK]D-FenderKatty: http://pastebin.com/BsJXYxMt
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15:00.08bsdiceoh, Katty.
15:00.16bsdicesup for V-Day :)
15:06.33*** join/#asterisk sekil (~sekil@78.24.104.73)
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15:29.29Katty[TK]D-Fender: sounds like quite the little line up!
15:29.33Kattywaves to bsdice
15:30.34Kattybsdice: heading north for the weekend, if the weather is agreeable
15:38.12*** join/#asterisk infobot (~infobot@rikers.org)
15:38.12*** topic/#asterisk is #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: Asterisk 11.7.0 (2013/12/17), 1.8.25.0 (2013/12/17); Standard: Asterisk 12.0.0 (2013/12/20); DAHDI: DAHDI-linux 2.9.0 (2014/01/30), DAHDI-tools 2.9.0.1 (2014/01/31); libpri 1.4.14 (2012/12/20) -=- Visit the official Asterisk wiki: wiki.asterisk.org -=- Please report bullying to an op
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15:54.20xaristaxHey someone know why the DTMF works with inbound calls but with externals goes down?
15:54.54[TK]D-Fenderxaristax: three is no such thing as "external" and "internal", and real details are required... and all but certainly debug and configs to match
15:55.32xaristaxD-Fender will be convinient to make an RTP Debug to see de problem?
15:58.07[TK]D-Fenderhold on that
15:58.17[TK]D-Fenderlets just see what you're even working with.
15:58.22[TK]D-Fender~pb
15:58.22infobotA "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.
15:58.24[TK]D-Fender^^^^^^^^^^^
16:00.55xaristaxThanks Fender this is my trunk
16:00.57xaristaxhttp://pastebin.com/rqy0Qimi
16:02.07xaristaxAnd this one is my extension
16:02.12xaristaxhttp://pastebin.com/nCG62eH4
16:07.31mirela666has anyone beed working with Mediatrix devices?
16:12.32[TK]D-Fenderxaristax: nat=force_rport,comedia <- providers are almost never behind NAT.  This should almost certainly be "no"
16:16.23*** join/#asterisk navaismo (~navaismo@201.124.114.174)
16:16.29xaristaxFender i do some test and i put in the extension dtmfmode=inband and the funny thing is that works
16:16.41xaristaxbut in the cli show me an error: Inband DTMF is not supported on codec g729. Use RFC2833
16:17.03xaristaxso my question is what the heck is using
16:17.44[TK]D-Fenderxaristax: user=XXX <- this parameter is not valid
16:18.03[TK]D-Fenderxaristax: and you have 2 "insecure" lines
16:18.30Kattyinsecures [TK]D-Fender's lines.
16:18.40[TK]D-Fenderindeed G.729 does not support inband (reliably)
16:19.07xaristaxfender XXX i put on purpose to hide my data :P
16:19.22xaristaxand i delete that line already thanks :D
16:19.28xaristaxbut what it works
16:19.33xaristaxit is a default for DTMF?
16:19.45xaristaxso inband is not supported but why works?
16:20.00Kattydebates playing with the polycom ftp directory bits today
16:20.14xaristaxi mean im happy that works but i dont like to see than warning on my cli and i wanna know what is using
16:20.18[TK]D-Fenderxaristax: ... that PARAMETER does not exist at all.....
16:20.35Kattyoh what to do what to do. i could put pretty pictures on my phones.
16:20.48Kattyor set ringtones that probably violate some kind of copyright law
16:21.04[TK]D-Fenderxaristax: I don't knwo what provider you're using, I don't see a call with SIP debug to see what they say they support, I also don't know you're using for that other peer.
16:21.06Kattymaybe make everything auto answer? decisions decisions
16:21.58[TK]D-FenderKatty: Do I what I did and set an entire department up with sync'd call-files pushing auto-answer & tt-monkeys on loop :)
16:22.34Kattyoh god.
16:22.39Kattythat sounds like fun ^___________________^
16:23.02[TK]D-FenderKatty: Was great watching 4 phones in a closed room all fire up at once :)
16:23.15WIMPywonders if Kattys nose got run over by a train
16:23.44[TK]D-FenderKatty: If it wasn't for lawsuit potential.... there's always recordings of gun-fire, breaking glass, and screaming....
16:24.03xaristaxi try to delete the rule dtmf from my extension and dont work again so the inband rule even when is not compatible with g729 is doing something
16:24.09[TK]D-FenderKatty: I'm committed to getting around to trying "Cards Against Humanity" this year....
16:24.11xaristaxwhat a weird thing
16:24.29Katty[TK]D-Fender: good. you'll enjoy it
16:24.53Katty[TK]D-Fender: there's some nice political jokes as well as crude humor
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16:37.31DevWork_hey peeps. good friday to everyone.  i had a question as was going to sleep last night. can you ring a group of devices, but only if they are not on an existing call?
16:38.19Kattyi can't see why not
16:38.34[TK]D-FenderDevWork_: Check if they are on a call.  Then choose the ones to dial
16:38.59DevWork_is there a command/app to see if a sip device is on call?
16:39.22[TK]D-FenderDevWork_: "core show function DEVICE_STATE"
16:39.43Kattyrefresh my memory, which file in the polycom ftp directory contains the path to the logo image thingy
16:39.48Kattywas that sip.cfg?
16:40.41DevWork_d-fender. Hmm as far as i know that only works via the console, i'd have to start up an external command with that. you know of a dialplan way?
16:41.10[TK]D-FenderDevWork_: that IS the dialplan way
16:41.31DevWork_Device_state(SIP/Device234) or something similar?
16:41.45[TK]D-FenderDevWork_: I just gave you the command to get the instructions....
16:41.53DevWork_Yea its on my console now.
16:42.09[TK]D-FenderDevWork_: go see what it gives you back
16:42.46DevWork_So I can use this with some branching logic awesome.
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16:46.14jeremy_lainedoes anyone here make heavy use of Bridge?
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16:49.24mjordanjeremy_laine: there are people that do, but that's probably not your actual question
16:49.45[TK]D-Fendermeta_questions--
16:49.51jeremy_lainemjordan: heh, right, am looking for people who manually bridge channels.. and encounter crashes
16:50.08mjordanjeremy_laine: what version are you running?
16:50.15jeremy_laine11.7.0
16:50.31mjordank, so if you're getting a crash when you run Bridge, that would be a bug. :-)
16:50.42jeremy_lainehave been struggling all day to get meaningful traces
16:51.11jeremy_laineI had previously tacked onto a bug I thought was related, and sent a (not fully complete) backtrace
16:51.34jeremy_lainebut over the past 24h I have been hitting the bug repeatedly
16:51.39jeremy_lainelet me pull up the old bug
16:52.27jeremy_lainehttps://issues.asterisk.org/jira/browse/ASTERISK-22875
16:52.53jeremy_lainemjordan: hey what do you know, you were the one who replied to me :)
16:53.54jeremy_lainemy crash still happens with the same sequence as the backtrace I sent (__ao2_find segfaults, being called via  ast_rtp_codecs_find_payload_code and bridge_p2p_rtp_write)
16:53.54mjordanSo your backtrace is different from David's
16:53.58mjordanright
16:54.05jeremy_laineyes, that was the conclusion, which is correct
16:54.08mjordanit's a masquerade yanking the channel out of an RTP native bridge
16:54.12mjordango ahead and open a new one :-)
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16:54.24mjordanwe've got a few other cases that are similar, but I'm not sure if anyone has actually gotten it reported
16:54.36jeremy_lainehm, is there any way to "anonymize" backtraces
16:54.45jeremy_laineas they contain phone numbers & such
16:54.46mjordannot really - you have to strip it out
16:55.05Penguinperl, sed
16:55.11jeremy_lainePenguin: nice :)
16:55.19mjordanKnowing that you used Bridge to cause it will help in figuring it out. That tells us that the channel got masqueraded - which makes it pretty easy to diagnose
16:55.25mjordan(or at least, easier :-) )
16:58.13jeremy_lainebasically the setup is two calls which are fired up via Originate
16:58.37jeremy_laine(one to an operator, one to a contact)
16:59.30jeremy_lainethe operator gets put into an extension which basically does Answer + Wait(1800)
17:00.04jeremy_lainewhen the contact picks up, he gets bridged to the operator
17:00.27jeremy_lainethe contact hangs up => the operator returns to the extension and waits some more
17:01.08jeremy_lainemjordan: and at some point things fall apart, and I start getting "bad magic number" warnings
17:01.30jeremy_lainemjordan: and eventually asterisk bombs, but always in bridge_p2p_rtp_write
17:01.49mjordanthat's good info - can you make sure that gets on a JIRA issue?
17:04.48jeremy_lainemjordan: am copying / cleaning up what I just wrote
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17:05.05mjordanjeremy_laine: thanks!
17:05.38*** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183)
17:05.49jeremy_lainehttps://issues.asterisk.org/jira/browse/ASTERISK-23310 , am going to add some debug info
17:06.47Greenlightp2p is the one where it avoids the asterisk core, right ?
17:11.03rrittgarnanyone have any tips on the errors in this polycom error log?  http://pastebin.com/azCdjj
17:11.10jeremy_laineGreenlight: yes, it seems to use reinvites
17:11.33Greenlightjeremy_laine: Oh.. I thought p2p wasn't direct media, more of avoiding the core
17:12.07*** join/#asterisk boratynskikamil (~kamilbora@wpb89.bialnet.pl)
17:12.20boratynskikamilHello, is it possible I see channels, but I don't see the signal then I call?
17:12.23jeremy_laineGreenlight: wait I may be mistaken
17:12.38jeremy_lainemjordan: added a backtrace
17:13.00GreenlightOnly reason I was wondering, I use a LOT of AMI briding in my app, but not see this issue, but since I record calls, it wouldn't do p2p, at least that's how I understood it
17:15.46boratynskikamilhttp://pastebin.com/v7Bf3TXm - dahdi_cfgh
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17:16.56boratynskikamilhttp://pastebin.com/TZwmmwuD - dahdi show channels
17:17.06boratynskikamilI should see connection when I call from-pstn, right?
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17:22.08jeremy_lainemjordan: ok have added some log messages along with the backtrace
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17:28.58boratynskikamilHm? Suggestions guys?
17:30.40WIMPyboratynskikamil: Maybe you can ask again. I don't understand your question.
17:31.14WIMPyBut your channelmap looks strange.
17:31.20boratynskikamilWIMPy: Yes, of course. I have 2 channels defined from-pstn, right? When I call, I do not see any log it is received. Is it possible?
17:31.31boratynskikamilWIMPy: You mean dahdi_cfg -vvvv?
17:32.06WIMPyyes
17:32.31boratynskikamilHm. I think it is strange too.
17:32.37boratynskikamilI generated it with dahdi_genconf, that's it.
17:33.04WIMPyHow many cards/ports do you have?
17:33.12boratynskikamilWIMPy: 2 cards.
17:33.36boratynskikamilWIMPy: http://pastebin.com/099JqfUP
17:34.51WIMPyInteresting, but I guess it can make sense then.
17:35.02boratynskikamilWIMPy: That's right.
17:35.14boratynskikamilAnd mainly... I am able to control GSM card, right?
17:35.28boratynskikamilBut... what is strange too? I don't see from-gsm channel. ;-/
17:35.30WIMPyI'd hope so.
17:35.40boratynskikamilBut 'gsm' comamand works well.
17:36.35WIMPyLooks like channel 1 isn't configured in chan_dahdi.conf.
17:36.50boratynskikamilWIMPy: But should it be?
17:37.07WIMPyOnly if you want to use it :-)
17:37.08boratynskikamilI am able to control 'gsm' card and I got [from-gsm] not found..
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17:37.28WIMPyWell, your dialplan is an entirely different matter.
17:38.00boratynskikamilWIMPy: Exacly, I think I should see connection... I mean, in verbose mode, see someone is calling...
17:38.23WIMPyOn the BRI, yes.
17:38.50boratynskikamilWIMPy: That's the point, where should I look for mistake? Would you like to see configs?
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17:39.35WIMPyWhich one? The missing channel? Well, that's obviousely ... missing.
17:40.11boratynskikamilWIMPy: Mayby not the channel. I thought about the thing, I am not able to see someone is calling. (:
17:40.43WIMPyTry to enable debug as well.
17:40.55WIMPyIf you still don't see anything, enable pri debug.
17:42.21boratynskikamilWIMPy: Still nothing.
17:42.26boratynskikamilcore set debug 9999
17:47.25*** join/#asterisk StaRetji (~vencuga@178.79.6.17)
17:47.42StaRetjiHi guys
17:48.20StaRetjiI've just noticed that .call files do not work for me anymore on asterisk 1.8 (was working yesterday)
17:48.29StaRetjichan_sip.c:20245 handle_response_invite: Received response: "Forbidden" from '"asterisk" <sip:asterisk@IPADDRESS
17:49.01StaRetjiIt must be some misconfiguration which I did by mistake, ,but I have no idea which one
17:49.18StaRetjioriginate from CLI gives same error
17:49.26StaRetjiany help is highly appreciated
17:49.37WIMPyWhere's the link between call file (not) working and someone rejecting your calls?
17:50.34StaRetjiWIMPy: wow, never thought of that
17:50.48StaRetjiit is gsm gateway
17:51.03StaRetjiso, this looks like it is not accepting calls?
17:51.15WIMPyThat's what it said.
17:51.35StaRetjimaaaan
17:51.42StaRetjilet me check it
17:58.49StaRetjiWIMPy: this is second time this week they you helped me
17:58.56StaRetjihow can I ever thank you?
17:59.39Qwell~thanks
17:59.39infobotbitte, Qwell
17:59.43Qwellmeh
17:59.48QwellJust send money...
18:00.08WIMPyMake money fast
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18:00.19pabelangerbitcoin cheap
18:03.31ipengineerIs there a way with pjsip to allow traffic without authentication from a specific host without dropping the secret from the ps_endpoints table?
18:04.16PenguinWith old SIP, that was done by using remotesecret instead of secret.
18:04.26Penguin(if I understood what you're saying you want)
18:05.02ipengineerPenguin: yea I am trying to set kamailio as a proxy in front of asterisk and it needs the secret from the ps_endpoints table but I want asterisk to accept all sip traffic from the proxy without challenging for auth
18:07.31*** part/#asterisk StaRetji (~vencuga@178.79.6.17)
18:08.20*** join/#asterisk JerJer[work] (~Burp@2620:106:e00f:f062:510c:54ac:b1f5:3843)
18:08.59JerJer[work]does anyone know if one can set the RDNIS on an outbound SIP call, which might get picked up by an ISDN PRI gateway??
18:09.05JerJer[work]Diversion header?
18:12.35WIMPyPossibly. You know that RTFM thing...
18:15.21*** join/#asterisk gusto (~gusto@2a02:810d:8600:8d4:21b:63ff:fe31:8426)
18:17.41JerJer[work]i have… i see a lot about receiving it, but so far nothing specifically saying one can set it
18:17.54JerJer[work]asterisk 11.7.0
18:18.25WIMPycore show function REDIRECTING
18:19.02WIMPyCALLERID(ani) also gets sent as diversion, which I guess is a bug.
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18:19.35monstercoWhat is your favourite open-source monitoring program for Asterisk? peers and servers (with alarm support)?
18:20.40mjordanipengineer: do you really need an authentication section for your pjsip endpoint then?
18:20.41monstercoI have been using Observium for a while and it's really nice but they have a hefty license fee for alarms (stupid)
18:21.00mjordanipengineer: you can still have a database table that Kamailio uses, but I'm not sure you need the endpoint itself to reference the authentication
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18:21.17ipengineermjordan: Yea we just added another column for password and set that to empty
18:21.35mjordanyou can also just have the endpoint not reference an auth
18:21.52mjordanfrom Asterisk's perspective, you want it to think there is no authentication for the endpoints
18:22.03JerJer[work]WIMPy:   coool… i had not found that and my google's for asterisk rdnis or  asterisk diversion hadn't matched anything 'redirecting'
18:22.03mjordanSo really, that password can live anywhere
18:22.08JerJer[work]WIMPy:  thank you
18:22.45mjordanin chan_sip, the password and the authentication was part of the peer (or endpoint in pjsip land). Since that's now broken up into two separate objects, you can decide whether or not Asterisk thinks the endpoint should authenticate at all, while maintaining the authentication for other things elsewhere
18:23.02ipengineermjordan: So the best way to achieve that I guess would be just to set the auth under ps_endpoints to null?
18:23.03mjordanipengineer: you can also noload the res_pjsip authentication modules :-)
18:23.11mjordanipengineer: yup
18:23.15ipengineerOk will do both of those
18:23.26mjordanif there's no authentication modules, there's no authentication :-)
18:23.53ipengineermjordan: makes sense to me.. We shall give it a try
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18:24.30WIMPyHmm. So if for some *** reason that module can't be loaded, you suddenly accept anything?
18:25.06ipengineermjordan: noload =>res_pjsip_authenticator_digest.so?
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18:43.43mjordanipengineer: should be
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18:45.33mjordanWIMPy: no, that doesn't mean anonymous access. That does mean that a request for an endpoint wouldn't be authenticated.
18:46.38WIMPySo all requests would be rejected then?
18:46.59mjordanWIMPy: nope. Endpoint identification (who a request is for) is separate from authentication
18:47.18WIMPyis confused now
18:47.39mjordanSo res_pjsip first figures out who a request is for.
18:47.56mjordanThat's endpoint identification. There are multiple ways that can be done, but in res_pjsip, everything gets identified for an endpoint.
18:48.02WIMPyThat's what I thought.
18:48.07mjordanAnonymous access has an 'anonymous' endpoint
18:48.38mjordanThat's separate from authentication. How you then authenticate for the endpoint you identified for is provided by one of the authenticator modules - of which, right now, there is only res_pjsip_authenticator_digest
18:48.45WIMPySo what happens if a non-anonymous endpoint is selected, but the auth module is missing?
18:49.05mjordanif you specific that it should authenticate but there is no authentication module, res_pjsip shouldn't have loaded.
18:49.10mjordaner, specified.
18:49.21WIMPyok
18:49.30mjordanBasically, however, if you don't ever want to authenticate - as is the case in ipengineer's scenario - there's no need to even have the module.
18:49.45mjordanIn fact, if you accidentally left in some authentication stuff, you wouldn't discover it until the endpoint couldn't communicate
18:49.47WIMPyThen I guess I read too much in to what you said to ipengineer.
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18:54.30WIMPyBTW (on the other topic) it would be great if someone could shed some light in to the meaning of all the CALLERID variables including REDIRECTING.
18:57.11mjordanWIMPy: I agree. Would you mind making a documentation bug for that? We really do need to clean that up.
18:57.25ipengineermjordan: So that worked great for us.. Is there way to suppress all of the warning though saying: No SIP authenticator registered. Assuming authentication is not required?
18:57.26WIMPyCan do
18:58.25Kobazmm, not really an asterisk question, but
18:58.47Kobazi'm trying to do a <a href="tel:555xxxetc,,,,1234#"> type link
18:59.05Kobazanyone know how do make a "click to call" link that inserts pauses and pound, etc
18:59.27Kobaz(that link doesn't work to that effect on android)
19:00.16WIMPyWell, an URI is not a dialstring. I think thats outside its definition.
19:01.57Kobazit might be
19:02.11Kobaz<a href="tel:555xxxetc">  works fine
19:03.34*** join/#asterisk biox (~jesseolso@173-165-253-93-minnesota.hfc.comcastbusiness.net)
19:03.59WIMPyDoesn't it say taht tel: URIs schould only contain international numbers?
19:04.52Kobazoh, maybe
19:05.04Kobazlike +1800555etc
19:05.20WIMPyHmm. Send_diversion=no does not seem to suppress diversion headers. Is that a bug?
19:05.43Kobazis that the right variable name?
19:05.51WIMPyThe official version is probably without the +
19:07.23WIMPyThat's what I find in the sample config.
19:08.19*** join/#asterisk tris (tristan@camel.ethereal.net)
19:09.39ipengineerDoes anyone know why the request URI in the notify is not the URI in the contact header from subscribe with pjsip? https://gist.github.com/zconkle/e2755a2cd7d82fea219b
19:10.25ipengineerThe request URI in notify has IP and Port of the SIP server not the contact from subscribe
19:15.05salz212hi, is there a possibility of using no certs for tls at client side?
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19:29.06vastina"Early to Early to Confirmed Dialog Forking" does that make sense to anyone in here?
19:29.15vastinaexcuse my ignorance if it's easy
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19:33.41WIMPySo, any comments on me seeing Diversion: headers with send_diversion=no? Or do I file another bug?
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19:55.28KattyWHAT THE ASTERISK SAY?!
19:56.20navaismopeee peee peee
19:56.36Kattyfatalerrorunknownsqlengine, noacceptableccompilerfound, libcppfailssanitycheck
19:56.45Kattydances with navaismo
19:56.51bsdiceKatty heading "north" like in US of A?
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19:56.57bsdiceKatty you like snow huh
19:57.04Kattyi /hate/ snow.
19:57.08bsdiceor is north more like San diego -> LA
19:57.44Kattyto me "north" is canada.
19:58.06pabelangerAnybody using a VXML interpret with Asterisk?
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19:58.45Kattyemphasises pabelanger
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19:59.15Kattysets pabelanger's emphasis level to strong
19:59.34bsdicenever been to canada
19:59.39bsdicetoo big
19:59.40Kattypabelanger: also, no.
19:59.51bsdicetoo many ways to die
19:59.53Kattythat's about the worst reason i've ever heard for not visiting Canada.
20:00.25Kattybsdice: in my opinion, 1 way is too many ways to die.
20:00.37bsdiceI am afraid of bears
20:01.05bsdicehey this Canada is off topic right, but since no one talks I figure its ok
20:01.13bsdiceif bored, please ignore me
20:01.14Kattyi always off topic.
20:01.37Kattybsdice: i very much doubt running into bears are a common every day occurance in canada.
20:01.48*** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
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20:01.49Kattybsdice: running into poutine daily might be more accurate.
20:01.56Kattybsdice: or running into hockey fans.
20:02.01bsdiceputin?
20:02.14Kattyno.
20:02.18Kattythat's different. also from russia.
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20:02.59Kattybsdice: look what i found! :> http://memecrunch.com/meme/Q08N/poutine-not-putin-h-t-sukitch/image.png
20:03.46bsdicedamn thats a lot of calories
20:04.17WIMPycalgaries?
20:04.22Kattymmm calories.
20:04.44[TK]D-Fender[15:01]Kattybsdice: running into poutine daily might be more accurate. <- Running & poutine are incompatible ;)
20:05.05Kattywhat if you're running /to/ the poutine?
20:05.12[TK]D-FenderOnly happens once ;)
20:05.13Kattyyou're certainly waddling away.
20:05.16[TK]D-Fenderyup!
20:06.58Kattyis there anything /useful/ to do with these polycom idle_display
20:07.09Kattyother than your company logo. which i guess is kind of useful.
20:07.33WIMPySend rude messages
20:07.42[TK]D-FenderKatty: I had live queue stats for 2 queues, 4 agents + 2 VM boxes
20:07.49Kattyyou mean cronjob something into the ftp directory to update the image?
20:08.26Katty[TK]D-Fender: how did you get the Idle_display to be live?
20:08.26[TK]D-FenderKatty: You could include things like a status page for things like day/night mode, server-based DND, calls missed, etc
20:08.37[TK]D-FenderKatty: set a refresh on it.
20:08.42Kattyoh.
20:08.44Kattyright.
20:08.50[TK]D-Fenderthat's the MicroBrowser, NOT the "idle image".
20:08.53[TK]D-Fenderit takes precedence
20:09.10[TK]D-FenderImage = boring... and you could use the MB to grab a flat image if you cared anyway
20:09.23Kattydoes the microbrowser require a certain minimum model of phone, or is that all firmware?
20:09.36Kattycause we have some /really/ old phones
20:11.01Kattythe most recent is uhhh...1.3.8 on this one
20:11.34[TK]D-FenderWhat model?
20:11.40[TK]D-FenderThat is beyond ancient
20:11.42Katty500 i believe
20:11.45Kattypossibly a 501
20:11.59[TK]D-FenderHeck IP 300/500/600 came with like 1.5.2 from factory...
20:12.05[TK]D-FenderCould be 2.3
20:12.16[TK]D-FenderIP 501 supports the microbrowser
20:12.31Kattyyou have a link to anything?
20:12.34Kattyinfobot: microbrowser
20:12.42[TK]D-FenderIP501 = 3.1.8 max
20:13.03[TK]D-FenderIt's about 6 lines in the XML
20:13.11[TK]D-Fenderstands out
20:13.40Kattyyes it's 3.1.9
20:13.42Kattyerr 8
20:13.51Kattyi'm all sorts of transpossy.
20:13.51[TK]D-Fenderonly ones that don't do MB really is 300/301/500
20:14.27[TK]D-Fenderthought the limited screens on some models seriously limits its usefullness
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20:19.43Kattyhmm. yes.
20:20.01Kattymb.idleDisplay.home
20:20.12Kattyi may tinker with that
20:20.30[TK]D-FenderIP32X/33X = 2x17 Char Matrix (nots sure on pixel usability)
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20:39.23jeremy_lainemjordan: ping?
20:40.37jeremy_lainemjordan: I see you marked the bug as a duplicate of #23312 (a posterior bug?) but I can't seem to access it (permission violation sayeth  Jira)
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20:47.48filejeremy_laine, it's not marked as a duplicate - it was cloned into another project
20:48.42jeremy_lainefile: ah my bad then, little terminology problem
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21:09.50starkshi, i have 2 questions. what version of asterisk runs on the digium switchvox pbx and what versions of asterisk are most compatible with cisco ip phones?
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21:12.57Qwellstarks: 1) Ask Switchvox, since it's commercial software.  2) The question you should be asking is "Which phones works better, since these Cisco phones are complete garbage?".
21:14.29starksi don't have the option...
21:14.38starks100+ cisco phones need to go on it
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21:21.29issackellyIf I run "Playback" in a dialplan does the next command run at the start or the end of the file which is played?
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22:02.41newtonrissackelly, dialplan execution continues after the application finishes
22:02.41DevWork_[TK]D-Fender, RE: faxing. I'm going to get an analog line from the phone company I think, and just do calls over voip. You mentioned sangoma, is there anything not priced as much as the B600? I've narrowed down my use case to where I want to ONLY send faxes, and just have hylafax queue them up and shoot them out.
22:03.40[TK]D-FenderB600 is a quad card.
22:04.03DevWork_Is that where the price comes from? I can get those modular ones I just saw they had 24 ports, sounded overkill.
22:04.54[TK]D-FenderA200 supports it as well and you can get a 2 FXO module for it, and no HWEC
22:06.21DevWork_So i need one fxs and one fxo module with no echo cancel
22:07.14DevWork_FXO for my analog in, FXS to the fax modem.
22:07.34DevWork_hardware echo bad for faxing so don't get it.
22:09.04DevWork_Well wait, I could just faxmodem out. sounds good.
22:09.11DevWork_Skip * at this point.
22:09.57DevWork_mostly wanted * for inbound, but if im not doing inbound I don't see a need for * for outbound... I dont know what I'd gain./
22:10.11DevWork_Hylafax can queue them out and send them at their leasure.
22:10.38DevWork_If I had a PRI, I would need * to use a channel on it from the faxmodem, but that probably not going to happen due to contracts with existing vendors.
22:24.09jeremy_lainemjordan: digging into some SIP traces, it looks as though the bug I reported might be triggered by a 500 error received during a RE-INVITE..
22:24.50*** join/#asterisk vlad_starkov (~vlad_star@88.226.34.183)
22:26.24jeremy_lainemjordan: one leg says "OK", the other leg fails to get reinvited and gets terminated => might this leave the remaining leg with a bogus view of where RTP should be going?
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22:31.11mjordanjeremy_laine: it shouldn't, but I'm curious if that occurs everytime with the re-INVITE, or if that just happened to show up when you did the Bridge AMI action
22:31.34jeremy_lainemjordan: I did not do an AMI action, it's the Bridge application
22:31.57mjordanjeremy_laine: it's the same under the hood, luckily
22:33.47jeremy_lainethe huge majority of reinvites work just fine, but what I saw today was:
22:34.33jeremy_laine- leg A (contact) calls the Bridge application to get bridge with leg B (operator)
22:34.43jeremy_laine- the reinvite for leg A and leg B are sent
22:36.36mjordanjeremy_laine: do you have a full debug log with 'sip set debug on' enabled?
22:37.07jeremy_lainemjordan: I have one which did not trigger a crash but did trigger "bad magic  number"
22:37.59mjordanyeah, please post that
22:38.02jeremy_lainemjordan: any chance of sending it privately to you ?
22:38.08mjordanbad magic number errors are badddd mojo
22:38.23jeremy_laine(IP addresses, phone numbers.. )
22:38.37mjordanHm. Alas, for someone to work it, it will need to go on an ASTERISK issue
22:38.45mjordanwe can lock the ASTERISK issue so that it is only visible by bug marshals
22:39.07jeremy_lainemjordan: yeah I took the "bad magic numbers" as "it didn't crash but could have"
22:39.11mjordanbut keep in mind I'm not the only one who fixes problems :-)
22:39.22mjordanthat's usually what it means.
22:40.33jeremy_lainehm the log is huge (40MB)
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22:40.46mjordanwe only need the relevant portion
22:40.53mjordanyou can strip out everything before those calls start
22:41.13mjordananywho, I have to bounce home. If you'd like to send it to me, I can create a locked down issue.
22:41.23mjordan(I'm assuming you can find my e-mail address :-))
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22:48.38issackellynewtonr thanks
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22:50.08jeremy_lainemjordan: have trimmed the log somewhat, bzip2'ed it and send it by email
22:50.52jeremy_lainemjordan: I included a comment with which two legs are of interest as far as I can tell
22:52.06mjordanjeremy_laine: thanks, I'll take a look. If it looks to be the same, I'll attach to the current issue after I lock it down. That way you can still see it - if I make it, you won't be able to.
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23:10.33JerJer[work]meep meep
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23:23.12DSTMum
23:23.14DSTMhey there?
23:23.36DSTMI'm new to IRC and got a funky asterisk problem, I was hoping I could pick someone's brain
23:25.38[TK]D-Fender~ask
23:25.38infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
23:26.10*** join/#asterisk biox (~jesseolso@173-165-238-157-minnesota.hfc.comcastbusiness.net)
23:26.45DSTMI'm running asterisk 11.7 with Queuemetrics on top for metrics, we log in and out of agents/extensions with it
23:27.23DSTMWe used to be able to put in any number into the extension box, as well as our extensions, and have asterisk ring it
23:27.55DSTMwhen we do that now it treats the number as an internal agent, and it won't dial it
23:28.18DSTMessientially I need to be able to have agents use whatever number as an extension
23:28.35DSTMnow it seems it's only letting them use predefined extensions
23:34.54[TK]D-FenderAnd how are you adding them?
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23:39.25DSTMadding what?
23:39.27DSTMThe extensions?
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23:42.29[TK]D-Fenderqueue members

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