IRC log for #asterisk on 20150807

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00:09.27moe`hey guys
00:09.47moe`having problems with getting linphone to bria calls working... anyone know of any magic there?
00:10.39moe`the call establishes, but no audio
00:11.09moe`now before we say audio devices, etc, we know these work on both ends cuz we can call via skype connect out to landlines and it works for both bria and linphone
00:11.20moe`its literally linphone <-> bria is the problem
00:13.21moe`[TK]D-Fender are you around dude?
00:14.11[TK]D-Fenderyup
00:14.27moe`any comments on the bria-linphone thing?
00:14.29[TK]D-Fenderand the answer for this is generally always improper network setup
00:14.41[TK]D-FenderStopping pinning names and start looking at DEBUG
00:15.23moe`ok but the same remote client, same box, running linphone ... no joy.  running bria... joy.  that's just a matter of how the client handles things then?
00:15.43[TK]D-Fenderyes
00:15.50[TK]D-FenderOne is making a WRONG assumption
00:16.02[TK]D-FenderOr compensating for your lack of properly configuring your server
00:16.18moe`ok, but when all remote clients are linphone it works
00:16.25moe`so bria is busted then, I gather
00:16.28[TK]D-Fenderno
00:16.45[TK]D-Fenderit may be COMPENSATING for your improper ASTERISK setup and covering your ass
00:16.51[TK]D-FenderStopp guessing and start looking
00:17.18[TK]D-FenderNo, bria is NOT "busted"
00:17.50moe`well I meant my bria config is busted
00:18.10[TK]D-Fenderno.
00:22.20moe`I find it odd that even on each box and the asterisk server itself, if we drop firewall, bria to bria works, linphone to linphone works, but bria to linphone not.
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00:22.33moe`looking at the asterisk logs, I see nothing that tells me why
00:22.38moe`but its not in debug mode
00:23.14[TK]D-FenderStop saying "odd" and "broken".
00:23.18[TK]D-FenderYou are jumping at excuses here
00:23.47[TK]D-FenderThere is no "odd".  There is nothing odd about this.
00:23.55[TK]D-FenderThese clients are not magical.
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00:33.49moe`there is something stupid I am missing here
00:34.24[TK]D-Fenderyes, 20 minutes later and still no debug
00:34.26[TK]D-Fenderthat's what's missing
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00:39.16moe`on "asterisk -rv"  ... "core set debug 10" should be verbose eh?
00:39.43[TK]D-Fenderuseless
00:39.46[TK]D-Fenderthat is not SIP DEBUG
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00:41.10moe`gotcha
00:41.55eduzimrsHi guys, im tryin to set up an * behind NAT with ISP failover, but i've set externip to ISP1, is it possible to turn this method automaticaly ? or every time i have to change the externip? i've been reading about externhost and NO-IP, is it the only way?
00:42.26[TK]D-FenderediIf you're on a changing IP there's that and STUN.
00:42.44WIMPyOr script something.
00:42.50[TK]D-Fenderyup
00:43.43WIMPySome (longer) time ago I used to rewrite the externip of sip.conf in ip-up.
00:44.28eduzimrsbut, using externhost to an FQDN and set externrefresh to the trick?
00:44.46eduzimrsbut, using externhost to a FQDN and set externrefresh to the trick?
00:44.51WIMPyIf that hostname is automatically updated...
00:44.52eduzimrsbut, using externhost to a FQDN and set externrefresh do the trick?
00:45.08eduzimrsyah, i'd put and noip client
00:45.22eduzimrsyah, i'd put a noip client
00:45.33eduzimrsyah, i'd put a noip client running each 30s
00:45.36WIMPyMight still take some time.
00:45.55eduzimrsthere is the time to update globaly
00:45.56WIMPyThey will blacklist you.
00:47.20eduzimrsyah i havent think that
00:48.07eduzimrsexternip does not support two ips right?
00:48.21eduzimrsthat doesnt sound good by the way
00:48.27WIMPyright
00:49.14eduzimrsso, there is no way doing that using * options
00:49.41WIMPyNot with chan_sip.
00:50.28eduzimrsright
00:52.13eduzimrsmaybe a script at cron using sed to substitute and sip reload
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00:59.07tompawEvening.
01:00.56WIMPyOr check if chan_pjsip can do better.
01:36.27moe`oh, got bria to one linphone client working
01:36.39moe`yeah [TK]D-Fender it is stupid settings, network
01:36.48moe`you da man
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05:47.39moe`I wonder how much fun it'll be to setup an inbound number via Skype for asterisk
05:48.17moe`got the outbound going, simply prefix 9 and out it goes
05:48.44moe`inbound could be fun, problem is skype offers numbers in areas I don't give a shit about
06:06.42drmessanoPrefix 9?  My god, why?
06:13.07moe`why not?
06:13.22moe`prefix 9 and whatever global number for outbound
06:14.14moe`hey, you're a yoda master of asterisk, is there a quick and dirty way to setup inbound and then once the call is connected the user must select an extension?
06:18.24moe`I can see multiple inbound numbers, but the plan is the same... the user must select an extension for a softphone.
06:19.04moe`that's it, that's all.  no secondary ringing, no other crap, just call the number, select extension, period.
06:19.47moe`if I could get that fired up I'd have numbers in a few countries
06:22.30moe`drmessano, file, malcolmd, any suggestions for that?
06:28.44moe`should I actually need SRV DNS records?
06:28.53moe`I think not, no?
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07:07.09Ricohi
07:08.21moe`ok I have inbound working routed to a specific extension, any suggestions on how to have it go to a "menu" where a user has to input an extension?
07:23.56ChannelZthat's the whole points of the dialplan
07:24.35ChannelZMake extensions in some context, then make the entry point for the calls Playback()/Background() a prompt telling them to enter something
07:24.40ChannelZthe rest is magic
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07:59.30moe`yeah the syntax for the conf files is what I'm after
07:59.35moe`I am impressed though
07:59.51moe`it was rather simple to direct an inbound number to an extension
08:00.03moe`asterisk++
08:00.36moe`I dunno if any of the asterisk developers are here, but if they are, dudes, I owe you some beers.  Damn good work.
08:01.03moe`the more I mess around with asterisk the more I am impressed
08:02.23moe`I started life in IT as a C programmer, so I at least I understand a little of what went into asterisk
08:02.39moe`thumbs up
08:06.14moe`ok now that I have an inbound, how do I setup so that users can call outbound appearing as from that same number?
08:06.19moe`asks the google machine
08:06.47moe`of course it has to do with my SIP uplink
08:07.34wdoekesmoe`: your users have a context from where they start a call. somewhere from there, you'd set CALLERID(num)=xxx
08:08.06moe`and in what wonderful config would that be?
08:08.34moe`I'm working with one context only at the moment
08:09.11wdoekesextensions.conf obviously, where you define the dialplan
08:10.49moe`grep tells me its possible to define callerid in sip.conf as well, assuming I guess that the upstream SIP provider accepts it
08:10.55moe`?
08:12.22wdoekeshah, the upstream sip provider, that's a matter of its own. there are lots of differences in how they want the callerid to be provided
08:12.57moe`in my case it's stupid skype connect
08:12.59wdoekesmoe`: as for the callerid= option, yes: that defines the initial callerid when calling from that sip device
08:13.36moe`in that case, if the SIP upstream will accept it, I could set that globally
08:13.40moe`I assume
08:13.41moe`?
08:13.49wdoekesI don't know about your particular ITSP, but you may need to tweak the sendrpid= option for that "device"
08:14.46wdoekesmoe`: callerid will not be set globally, but you could use a template from which all your sip accounts inherit
08:14.46moe`right now, so far, I'm only plugged into skype
08:15.09wdoekesbut more common is to do it from the dialplan, just before you Dial(SIP/your-skype-itsp/${EXTEN})
08:16.10wdoekesor, if they take callerid in the From, you could cheat and set it as the fromuser= in the [your-skype-itsp] device config
08:16.23moe`ah, ok, so in my stupid case I have 9 prefix to use skype outbound, so using a different prefix I could adjust it to appear to come from different source(s)
08:17.38wdoekesI don't know what you mean by prefix
08:18.26moe`oh.  when a client connects to my asterisk, they dial 9 and then the global number to get "out"
08:18.56moe`but my SIP link with skype, they're being cunts on the caller ID
08:19.04moe`must verify/register my company
08:20.10moe`my company is unixninja.net, you bastards  :)
08:20.14wdoekesah, like that. yes, you could set the CALLERID based on the that prefix. (e.g. ExecIf($["${EXTEN:0:1}"="9"]?Set(CALLERID(num)=123); Dial(SIP/your-itsp/${EXTEN:1}) )
08:20.42wdoekess/ the that / that /
08:24.00stefan27Any new educated guess when 13.5 goes from rc to release?
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08:28.33moe`not understanding the expectations between 13.4 and 13.5, can I ask why?
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09:02.53afournierhi
09:06.36afournieri had problem in the past when loading asterisk "compiled later" modules. i.e building asterisk once without the module, installing it, cleaning the sources, rebuilding asterisk again with the module and then only installing the module, asterisk won't load it, until a complete install is redone... is it normal ? is it a known problem or security feature ? is it due to some compilation timestamp ? is there a way to bypass this ?
09:09.33afournierin fact, it's not really accurate, at both compilation time the module was built, but only installed the second time, menuselect.makeopts did not change from first compilation to the second
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09:24.37afournierit looks like /usr/sbin/asterisk and e
09:25.14afournierit looks like /usr/sbin/asterisk and /usr/lib/asterisk/<module_name_here>.so all contain the same md5 :)
09:27.59afournier./build_tools/make_buildopts_h:BUILDSUM=`echo ${BUILDOPTS} | ${MD5} | cut -c1-32`
09:28.28afournierso it means BUILDOPTS changed during these compilations :/
09:29.22afournierfair enough
09:29.35wdoekes:)
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14:06.40Juniorhello!
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14:07.47JuniorFor linux (ubuntu 14.04) what piece of software (free) should i use for my sip accounts? (I need to be able to import contacts from a csv list and zoiper free doesn't do that). Thanks!
14:08.33JuniorMeaning if anyone is willing to recommened something as i don't want to chose a wrong software.
14:19.20mirela666Junior, I can recomend, Jitsi, Linphone, Twinkle
14:28.32Juniormirela666, thanks! I have downloaded jitsi and will continue with the rest
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15:08.59hexanolI have a question about CDR in Asterisk 13
15:09.04hexanolIn a scenario "Alice calls Bob then Bob answers", but where the call is setuped by an originate instead of Alice dialing Bob's extension
15:09.12hexanolThe "billsec" in the resulting CDR includes the time Bob phone was ringing
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15:09.18hexanolIs this the expected behaviour ? Is this behaviour undefined ?
15:26.51mjordanbillsec is from when the channel was answered. Period. It's based on the actual change in the channel state, so I'm not sure how that would be possible.
15:27.05mjordanIf you think that is happening, I'd file an issue, but make sure you get logs
15:27.10mjordanincluding with 'cdr set debug on'
15:27.22mjordanbut I'm pretty sure we're going to find out that the channel was answered when you didn't think it was
15:30.18hexanolthe channel is answered indeed
15:30.26mirela666hexanol, in the dialplan you are probably doing Answer() as within first step
15:30.31mirela666steps*
15:31.01mirela666that is setting the flag on channel than time for calculating billsec
15:31.29mirela666hexanol, remove the Answer and leave the Dial application to set that flag on actual answer
15:31.44hexanolthere's no explicit Answer in my dialplan
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15:31.55hexanolthe behaviour comes from the fact I'm using an originate
15:31.56mirela666hm
15:32.16hexanoli.e. if Alice calls Bob by dialing Bob's extension
15:32.20mirela666then you might pastebin the dialplan
15:32.38hexanolthan billsec is the time the conversation between Alice and Bob
15:32.42hexanolbut if I do an originate
15:32.56hexanoli.e. "channel originate SIP/alice extension 1022@default" (1022 is Bob's extension)
15:33.05mirela666oh
15:33.12hexanolthen billsec is the time the conversation between Alice and Bob + the time Bob's ring
15:33.24hexanolbecause indeed, Alice's channel is answered... when she answers the originate
15:34.15mirela666I'm not sure you can fix it with originate
15:34.28mirela666maybe someone else has more exp with this
15:35.11hexanolwhat is a bit unfortunate is that in these case you can't really use the billsec
15:35.45hexanolto know how much time the conversation between Alice and Bob lasted
15:38.37hexanolthat's why my question was "is this the expected behaviour" -- if it is, that's ok with me
15:38.44hexanolif it is not, then I could open a ticket on the tracker
15:39.20mjordanit is expected, as billsec is defined as being the time from when the Party A channel is answered to when the relationship between the Party A and Party B channel is dissolved for that CDR
15:39.44hexanolalright
15:39.48hexanolthank you for the explanation
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15:55.34hexanolthere might be a small error in the "Blind transfer" example of the CDR specification
15:55.52hexanolin the second CDR, the duration and billsec values are not equal
15:56.34hexanolbut Alice's channel was already answered the time SIP/charlie was called
15:57.07hexanoli.e. the start and answer value for the second CDR should also be equal (that's also what I have if I do the test)
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17:49.10dwarf_lonaHi guys, need some help on asterisk queue
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17:50.07dwarf_lonaSIP/5073 (dynamic) (On Hold) has taken no calls yet
17:50.37dwarf_lonawhat does On Hold means?
17:54.22robmalIt means the member is paused.
17:55.14dwarf_lonaHi robmal, is it (paused) when the member is on pause?
17:55.36dwarf_lonamy problem is i cant change the status of the membe
17:55.38dwarf_lonamy problem is i cant change the status of the member
17:55.52dwarf_lonai tried to remove and add the member
17:55.58robmalqueue unpause blahblahblah
17:56.01dwarf_lonai also tried to pause and unpause the member
17:56.23dwarf_lonai also tried to check the channel and hangup but no avail
17:56.37dwarf_lonaany thoughts?
18:04.34Milencowhat asterisk version are you using/
18:04.44Milenco@ dwarf_lona
18:05.03dwarf_lonaasterisk 10.9
18:06.06[TK]D-Fender10 = dead
18:06.10[TK]D-Fenderupgrade immediately
18:06.16[TK]D-Fender12 = dead.  Don't go there either
18:06.58pjensen00If you tell me that 13 is dead I'm doing to die too
18:07.10Milencoanyway dwarf_lona, from what i read the member reaches on hold state when it's in a call but that call is paused/on hold
18:07.18Milencopossibly because of a warm transfer
18:07.30Milencoeither use asterisk 11 or asterisk 13
18:07.39dwarf_lonais it an issue on 10.9?
18:07.48Milenconot sure, or even likely
18:08.11Milencobut its always hard to support outdated software versions, because many things got fixed in the newer ones..
18:08.17dwarf_lonaits just weird coz it doesnt happen before.
18:08.34dwarf_lonanow it happen in random
18:08.35Milencoi'm still using asterisk 1.8 here and got weird bugs as well :(
18:08.48Milencohard to believe its totally random
18:08.59dwarf_lonathere are also times that after putting the call on hold, the calls hangup
18:09.02Milencoits possibly, but it usually should be reproducable
18:09.08dwarf_lonabut that doesnt happen on all agents :(
18:09.20dwarf_lonayeah, me too
18:09.31dwarf_lonai tried to replicate the issue, but it doesnt happen to me
18:09.50dwarf_lonatried putting the call on hold for 2min, 5mins, and even 15mins
18:09.52Milencocan the problem be related to the phones themselve?
18:09.57dwarf_lonabut I was able to go back to the call
18:10.06dwarf_lonai was thinking about that also
18:10.07Milencoso whats the issue exactly
18:10.13dwarf_lonathey are using the phones Hold button
18:10.44Milencoyou mean you put a new member in the queue and it enters with the on hold state?
18:10.48Milencowithout an active call?
18:11.00Milenco(from the phones perspective)
18:11.26dwarf_lonafirst - when agent is engage on a call and put the cx on hold, when they unhold, call is already dead
18:11.49dwarf_lonasecond - when they join the queue they are already (on Hold) state
18:11.59dwarf_lonawithout any active call
18:12.03*** join/#asterisk spicyramen_ (~Adium@173.227.7.2)
18:12.13dwarf_lonaSIP/5073 (dynamic) (On Hold) has taken no calls yet
18:12.23Milencodid you verify there isnt a call still going on?
18:12.36dwarf_lonachecked on active channel, and there's none
18:12.53Milencoyou can manually hang up these stuck calls with 'channel request hangup'
18:13.07dwarf_lonathat's actually what i do
18:13.24dwarf_lonabut the thing with this particular problem is there no active channel to hangup
18:13.40dwarf_lonaso im stuck with (onHold) state
18:13.47Milencoif asterisk doesnt see the channel and still shows the agent in 'on hold' state then its probably an asterisk bug
18:13.59Milencoalthough my knowledge is very limited...
18:14.11dwarf_lonaunderstood
18:14.17Milencothis is a related issue from 2007
18:14.17Milencohttps://issues.asterisk.org/jira/browse/ASTERISK-10617
18:14.23dwarf_lonahow about on my first issue, any thoughts?
18:14.30dwarf_lona<PROTECTED>
18:14.45Milencoyou should check with sip debugging on
18:14.47dwarf_lonabut the channel is still active
18:14.58Milencoand see whats happening, although it can be a little hard to interpret
18:15.16Milencoeasier might be to just check the verbose log for that call
18:15.16dwarf_lonayes haha
18:15.33Milencoinside asterisk config or in message logfile (if you have verbose logging to file specified)
18:16.24Milencothe first issue reeks for sip/network issues
18:18.10dwarf_lonamy setup is - asterisk server is on a colocation. SIP users are on a remote location behind firewall
18:18.31dwarf_lonaasterisk -- internet --- firewall --- users
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21:15.22Cruz4prezIs there a good recent comparision of AsteriskNow vs FreePBX distribution?  All I'm finding is stuff thats several years old.
21:16.06robmalEveryone using GUIs died in the process.
21:51.24*** join/#asterisk roler (~roler@unaffiliated/roler)
21:53.09*** topic/#asterisk by mjordan -> #asterisk The Open Source PBX and Telephony Platform (asterisk.org) -=- LTS: 13.5.0 (2015/08/07), 11.19.0 (2015/08/07), 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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22:09.46*** join/#asterisk spicyramen_ (~Adium@192.195.80.114)
22:10.52spicyramen_hi people, Im using  Asterisk 13.4.0 and getting Failed to set remote answer sdp: Called with SDP without ice-ufrag and ice-pwd. using SIPML5
22:10.58spicyramen_any ideas?
22:13.01spicyramen_icesupport=true is true in rtp.conf
22:28.23*** part/#asterisk mjordan (mjordan@nat/digium/x-bdevekcfsufjzasw)
22:38.23spicyramen_stun show status showing (null)                    0     30      3        INIT    0.0.0.0          0
22:48.15WIMPyCruz4prez: There's nothing to compare any more.
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22:52.44Cruz4prezWIMPy are you saying they are the same?  Freepbx people say no with an overly vague answer saying they have no idea what asterisknow/digium does
22:52.51*** join/#asterisk johnny_|_ (~johnny@unaffiliated/johnny-/x-2623418)
22:53.39WIMPyAsteriskNOW is now FreePBX only. The Asterisk GUI is dead as a Dodo.
22:54.36Cruz4prezRight, I know asterisknow uses freepbx, but what I'm asking are what other differences are there?  what other packages?  what other integrations?
22:55.27WIMPyI don't think that peole askin such questions are the target of either one.
22:56.14*** join/#asterisk superscrat (~asanders@173-21-89-217.client.mchsi.com)
22:56.51WIMPyThere was a rumour that the both distros are actually identical, but I don't know if that's true.
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22:58.22Cruz4prezyeah, many people seem to think that, but freepbx definitely seems to deny it.  I know the freepbx iso is about 100 meg larger
22:58.36robmalBitmaps.
23:03.46spicyramen_my dear asterisk people, I just opened all udp and tcp in my asterisk 13.4 any idea
23:04.06spicyramen_why stun show status shows: Hostname                  Port  Period  Retries  Status  ExternAddr       ExternPort
23:04.06spicyramen_(null)                    0     30      3        INIT    0.0.0.0          0
23:05.30WIMPyFree calls
23:05.42WIMPyNo working STUN server?
23:06.14spicyramen_<PROTECTED>
23:06.14spicyramen_Module                         Description                              Use Count  Status      Support Level
23:06.14spicyramen_res_stun_monitor.so            STUN Network Monitor                     0          Running              core
23:06.31spicyramen_thats correct, it shows: ip-172-31-51-6*CLI> stun show status
23:06.31spicyramen_Hostname                  Port  Period  Retries  Status  ExternAddr       ExternPort
23:06.31spicyramen_(null)                    0     30      3        INIT    0.0.0.0          0
23:07.40WIMPyI've never used the stun module, so I don't know what it tell, but to me that looks like it was unable to contact any STUN server.
23:10.18spicyramen_nvm I configured res_stun_monitor.conf
23:10.21spicyramen_and now works
23:10.25spicyramen_sorry user error
23:10.32spicyramen_I though rtp.conf will be enough
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