IRC log for #asterisk on 20150731

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06:36.58*** 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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07:57.47nunneAnyone had problems with icesupport=yes in sip.conf and rtp.conf and all of a sudden Asterisk stops accepting incomming SIP channels all together after a while. I can't see anything in any logs. It just stops right up. Really weird.
07:59.53Chainsawnunne: Normally suggests your STUN server has stopped accepting connections.
08:00.00Chainsawnunne: Do you host that yourself or are you using someone elses?
08:00.16nunneChainsaw: using googles. but will have a look at it. thanks.
08:01.12nunneChainsaw: but will remove it from my sip.conf general. I only need it for webrtc. and will install my own instead. maybe I flooded googles :P
08:01.49Chainsawnunne: It may be an automated countermeasure on the Google server, yes. It's all speculation unless you capture the packets and compare a good run to a bad run.
08:03.08nunneChainsaw: thanks for pointing me in the direction. I will have a look at it. But sounds like a plausible thing they might be doing. Specially since it might be over 100 call setups per second. So might as well setup a stun server myself to save on some network traffic.
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08:04.24Chainsawnunne: "stuntman" is a decently maintained STUN server codebase.
08:04.31Chainsawnunne: Most of the others appear to be long abandoned.
08:05.45nunneChainsaw: thanks
08:12.41yun1989hello every one, I need when I receive one call to "1099", generate one call from 1099 between 343
08:13.14yun1989I try to use the command Originate but i don't have sucess
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08:46.52Ice_StrikeIs there public log of this channel?
08:47.03pjensen00Don't think so
08:48.19ChainsawSome people log it and put it online actually.
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08:50.51Ice_StrikeFound it lol https://botbot.me/freenode/asterisk/
08:51.06Ice_Strikereal time log too, nice.
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09:33.33yun1989hello every one, I need when I receive one call to "1099", generate one call from 1099 between 343
09:33.39yun1989<PROTECTED>
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09:42.03Jason_weileileioh ,i  am  come in.
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09:44.12Ricois there a way to tell asterisk 13 (pjsip) tu add a route header four outgoing requests ?
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09:47.41Ricos/four/to
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09:58.31Jason_weileileihi
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10:23.24pjensen00hello
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11:20.38cyford33hi
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12:34.41johnny_|_Hi, if server responds with WWW-Authenticate: Digest nonce="1400543250:ab9fbe93a438d3bd72fd1b7553cc2847",algorithm=MD5,realm="sip.someprovider.net". Does that mean use 1400543250 nonce to get ab9fbe93a438d3bd72fd1b7553cc2847 response?
12:37.46fileno, the nonce is used as-is in the response generation
12:38.40johnny_|_thanks, I omitted the fact that it might not be asterisk, sorry
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12:40.13filethe way it works is a defined way, the content of the nonce is opaque
12:40.19fileif it has meaning then that's specific to the implementation
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12:57.20tompawMorning
12:58.03tompawGuys, I need a favor. Can someone please do a lsof | wc -l on their asterisk user (non-root) and report the results together with number of users using the pbx atm?
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13:53.19tompawres_pjsip_pubsub.c: No registered publish handler for event presence << what's this?
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13:55.50Katty[TK]D-Fender: you are getting old.
13:55.53Katty[TK]D-Fender: but i still love you.
13:56.47[TK]D-Fender<3
13:56.48tompawCan I disable presence with pjsip?
14:00.32MaliutaLaphey Katty
14:00.52Kattywaves to MaliutaLap
14:02.09MaliutaLapjust went to the most amazing jazz gig in ages
14:02.25MaliutaLapbetter than Herbie Hancock and Chick Corea the other month
14:04.05MaliutaLapMy mate was recording stuff on his phone - should have a bootleg mixed in a couple of days
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14:07.43KattyMaliutaLap: very nice. what sort of jazz?
14:08.57Kattywe talking like big band swing?
14:09.18MaliutaLapKatty: wild, mad jazz - the kind that should be played somewhere like the Blue Note club where audience members aren't shushed for tapping beat on their legs, or can yell appreciation, and not it a concert hall where the old people suck the fun out of it
14:09.18Kattyor miles davis
14:09.32MaliutaLapJames Morrison with Megan Washington
14:10.06Kattythat sounds lovely ^_^
14:10.14MaliutaLapand don't mention davis around me - the man was the biggest thief in the jazz world, always took credit for other peoples music :)
14:10.37MaliutaLapthe pianist was so good I expected them to pull out some Monk
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14:16.30MaliutaLaphehehe I have coined the term "Megasm" it's like an orgasm, but comes from listening to Megan Washington perform
14:16.35MaliutaLapand it's catching on
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15:29.54Ricobon week end !
15:34.40MaliutaLapRico it's 01:35 on a saturday - I have had a head start on you :)
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16:13.54x86_64test
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16:24.07florian333444_hello
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16:27.55florian333444_question: (Asterisk 11)  i need to IGNORE 1 msn. i tried to put a list into /etc/asterisk/chan_dahdi.conf   in "msn=123,234,345,..." .... unfortunately this list may not be long enough, i needed to list 17 msn because i have 2 isdn-lines. ideas ?
16:28.24florian333444_the 18th msn is the one id like to ignore, thu
16:30.46florian333444_up to 6 or 7 msn are fine, but say msn 8-17 will be ignored (not in the msn-list ...) too
16:31.56florian333444_wanna put it more simple: what is the easiest way to tell asterisk to ignore a specific msn ?
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16:41.32rmudgettflorian333444_: The msn list in chan_dahdi.conf is per ISDN span and is only for when you have that span configured as PTMP CPE mode.  The msn list string can be up to 79 characters.  When used, Asterisk will ignore incoming calls with a callerid not matching a msn in the list.
16:42.19florian333444_rmudgett thx.
16:42.41florian333444_rmudget: but my msn-list is 17 msns long and far longer than 79 chars
16:43.15florian333444_i tried wildcards in there like "*" or "X"  to shorten my numchars.
16:43.20florian333444_but that didnt work
16:43.40florian333444_rmudgett: ideas ?
16:44.52rmudgettWhy do you need so many MSNs?
16:45.09florian333444_rmudgett its a small company
16:46.20florian333444_irmudget: i just need asterisk to completly ignore ONE specific msn ....
16:46.28florian333444_*rmudget
16:46.31rmudgettThat list is to allow other devices on a multidrop configuration to share the line so Asterisk will not attempt to steal calls from another device.
16:46.56rmudgettIf you don't have any other device on the line then you don't need to use the list.
16:47.15florian333444_ok i understand
16:47.28florian333444_mmm i have a fax-device on that line
16:47.34florian333444_but its not fax, its edi
16:48.14florian333444_so i cannot use forward-fax within elastiks because it doesnt recognize edi as fax
16:48.37florian333444_so i want asterisk to ignore ONE msn
16:49.02florian333444_*the protocol used is not fax, but edi
16:49.39florian333444_rmudget: do you have an idea howto tell asterisk to ignore ONE msn =
16:49.42florian333444_*?
16:51.00rmudgettWhen that list is not empty it tells asterisk which calls to accept.  If it's not listed it won't respond.
16:51.02florian333444_if the msnlist was up to say 250 characters i was fine, but its 79
16:51.17rmudgettThere isn't an option to do the opposite.
16:52.01florian333444_can i use wildcards inside chan_dahdi.conf ? if so, could you tell me how than looks like ?
16:52.53florian333444_...or msn-abreviations or so ?
16:54.18florian333444_...or tell 2 lists, each one for a span ?
16:57.06rmudgettflorian333444_: Just looked at the code and the list can be made up of dialplan extensions including dialplan patterns "_NXXNXXXXXX"
16:57.32rmudgettThat should work for you
16:58.12florian333444_ahhh...
16:59.32florian333444_so    msn="_2X3X,_4XX44" and so forth ?
16:59.41rmudgettyes
16:59.55florian333444_cool ... so cool i check it right now off course !
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17:01.26rmudgettHah.  It was even stated in the chan_dahdi.conf.sample file.
17:01.29rmudgett; Here you can give a comma separated list of numbers or dialplan extension
17:01.29rmudgett; patterns.  An empty list disables MSN matching to allow any incoming call.
17:01.47florian333444_really ? oh so bad i missed that im sorry
17:02.02florian333444_would you be so kind an paste that line here ?
17:03.55florian333444_rmudget: i think i did it............. HOORAYYY ! thx a lot !!!!!!
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17:09.21florian333444_thx so much ! a bunch of flowers to your room !!
17:09.24florian333444_cu !!!
17:11.30ryaoI have never used Asterisk. At the moment, my home has an Obi100 ATA connecting the POTS phones to Google Voice as well as a fax machine. Would it be possible to install a FXS into a supermicro system running Open Indiana, setup Asterisk to connect to Google Voice and then be able to receive and send regular calls as well as faxes using something like mgetty or hylafax{,plus}?
17:25.22cyford33is there a tool to check dialplans for errors
17:29.26filenot really
17:29.39filesyntax stuff will be output on load
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18:22.10pjensen00Switch to ARI.  Come to the dark side!
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20:20.42JimDickensonI am trying to get asterisk 13 going and VM_INFO seems to not be present
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20:21.41johnny_|_Hi. regarding https://issues.asterisk.org/jira/browse/ASTERISK-25265 issue. I've raised this issue here one week ago. To recap: calls _to_ firefox jssip client fail. Now I've applied this * 13.1 patch to * 11.18.0. It's still not working (I'll post logs) but let me just ask this to make it clear. Can I make calls to firefox using SRTP media encryption (codec is opus and vp8 passthrough) or do I _have_ to
20:21.43johnny_|_use DTLS?
20:24.44JimDickensonHow does one find out if a mailbox exists in Asterisk 13?
20:27.31[TK]D-Fendervoicemail <tab> from cli
20:27.52[TK]D-Fenderand there are dialplan functions for that at that level as well
20:28.31rmudgettVM_INFO still exists.  Is app_voicemail loaded?
20:41.07JimDickensonAs far as I know. VMCOUNT is present but not VM_INFO. It does not show up in core show functions nor when I try to use it.
20:42.47rmudgettVMCOUNT is supplied by a different module.
20:43.02rmudgettVM_INFO is supplied by app_voicemail
20:50.43JimDickensonThanks. I did not have app_voicemail.so
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21:15.28rexwin_is it okay to stop fail2ban-server as it taking up a lot of resources?
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22:21.12Ice_StrikeIf a call center have 500+ agents for outbound calls and put all in 1 astrisk box is asking for trouble. What is your solution?
22:22.00WIMPyDidn't you find the answer to the 2nd part yesterday?
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22:23.55Ice_StrikeYesterday?
22:24.04Ice_StrikeI don't remmeber about that answer
22:24.09Ice_StrikeI will scroll up
22:24.41cyford33probablly was dont put all in 1 box
22:26.46Ice_StrikeWIMPy yea yesterday was about ARI - creating own Queues application to seperate server and then scale to mulitple servers. Yea that would work.
22:30.44[TK]D-Fenderyou said "outbound calls"
22:30.49[TK]D-Fenderhow does taht relate to Queues?
22:31.24WIMPyOptimizing bad impressions :-)
22:31.46WIMPyOr sometimes getting your line cut by some watchdog.
22:33.30Ice_Strike[TK]D-Fender Yea I got mixed up, my bad.
22:34.18[TK]D-Fenderand you haven't clarified it either
22:34.24[TK]D-Fenderso which is it
22:34.37[TK]D-Fenderbecause 500 calls through 1 decent server isn't ahuge deal yet
22:37.08Ice_StrikeSomeone said to me yesterday that 500 agents on one host require very complex system(like kamailio with custom code). not doable on asterisk.
22:39.39[TK]D-Fenderhttp://www.voip-info.org/wiki/view/Asterisk+dimensioning
22:39.56[TK]D-Fender"someone said" sounds like vague crap.  Especially since your needs are never expressed properly
22:45.12Ice_Strike[TK]D-Fender Remember diagram I show you few days ago? Basically Originate a call, Caller Answered, AMD Check Passed and then passed to the Queue
22:45.21Ice_StrikeUsing AMI to do all that
22:45.42[TK]D-FenderHow many calls SITTING in the queue at a time?
22:47.15cyford33[TK]D-Fender  i found the reason my server goes bizzark after using busy()....................      if i dont answer the call on a 1800 number an send a busy the networks lcr and re route the calls ...    so with 1 call i end up with 4 calls to hangup
22:47.29*** part/#asterisk kharwell (kharwell@nat/digium/x-vopkwxsmkowuvkcj)
22:48.10WIMPyThat still doesn't explain the difference to using Hangup.
22:48.11Ice_Strike[TK]D-Fender It is assumption and scenario. Let say 300?
22:48.20cyford33i noticed when it kepted originating on same priority it was using differnt channels
22:49.14[TK]D-FenderIce_Strike, so 300 actually WAITING for agents?
22:49.17cyford33someone said busy() and conjustion() did something to the channel
22:49.21[TK]D-FenderThis is horrible
22:49.33Ice_Strike[TK]D-Fender Oh lol. No
22:49.36cyford33<PROTECTED>
22:49.48Ice_Strike[TK]D-Fender Let say max 50
22:50.02[TK]D-FenderIce_Strike, so use that link as a reference
22:50.10Ice_Strike[TK]D-Fender 500 agents is already taken and talking to customers.
22:50.18Ice_Strikeok
22:50.20[TK]D-Fender"ready " is not a LOAD
22:50.23[TK]D-FenderLOAD = WORK
22:51.48*** join/#asterisk theron (~theron@173-20-126-202.client.mchsi.com)
22:51.53Ice_StrikeAh ok
22:52.50*** join/#asterisk lvlinux (~ruel@unaffiliated/lvlinux)
22:52.52cyford33<rmudgett> Busy() and Congestion() are not meant to be used in the h exten.  Both of those applications indicate to the channel driver a busy/congestion signal and wait forever for the channel to hangup.
22:53.34WIMPyYes, but it's not being used in h.
22:53.49Ice_StrikeCan Asterisk queues() use external database for storage?
22:54.17cyford33yes,  but as soon as my 1800 number provider gets the signal they reroute
22:54.48cyford33if i use hangup.    py provider keeps getting a reinvite  and keeps the line active
22:54.59cyford33for 20 secs  or 3-4 retries
22:55.17WIMPyA reinvite after hangup? That doesn't make sense.
22:55.24cyford33or invite
22:55.37Ice_StrikeAh just reading http://manuals.loway.ch/QM_UserManual-chunked/ar01s19.html
22:55.41cyford33<PROTECTED>
22:55.44Ice_StrikeStoring queue data on MySQL
22:55.46WIMPySame
22:56.26*** join/#asterisk theron (~theron@199.201.65.2)
22:57.04Ice_StrikeMight be useful for asterisk queue failover destination?
22:57.27cyford33so all senerios end up multiple invites if not answering the freaking 1800 number...   but if i use hangup()  the line never ends till the last one finnishes...         and if i use busy()  the line ends but the same call comes in like 3-4 times
22:57.49*** join/#asterisk wonderworld (~ww@ip-176-199-166-61.hsi06.unitymediagroup.de)
22:58.43cyford33if i answer and hangup()  its good,   or answer and busy() its good..   but then i get hit with 85 cents payphone fees lol
23:59.36*** join/#asterisk fstd (~fstd@unaffiliated/fisted)

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