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| 01:02.49 | carrar | moo |
| 01:08.42 | igcewieling | I love you Adtran! |
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| 04:29.22 | gladier | hey guys - what do you use for joining people to your sip video conferences? |
| 04:29.46 | [TK]D-Fender | gladier: Asterisk does no such thing |
| 04:30.06 | [TK]D-Fender | gladier: The most we have is "follow the speaker" in app_confBridge |
| 04:30.42 | [TK]D-Fender | gladier: * doesn't re-encode video in any way |
| 04:31.01 | gladier | not exactly what i'm talking about - what i'm looking for is a solution i can place as a "sip endpoint" of sorts that allows people to join my sip video conference via a web browser - similar to the vidyoportal/router/player combintation just not in a 50k price bracket |
| 04:32.12 | [TK]D-Fender | gladier: what does Google-ing up "sip video web client" show so far? |
| 04:32.25 | [TK]D-Fender | gladier: There are probably Java apps for this |
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| 04:55.19 | gladier | logitech lifesize is apparantly the fit |
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| 07:24.11 | deo | guys need some help.. i dont whats the meaning of this in my console.. |
| 07:24.48 | deo | Channel Context Extension Prio State Application Data CallerID Duration Accountcode BridgedTo |
| 07:24.49 | deo | IAX2/to-comp2 from-internal STARTMEETME 1 Up MeetMe 2703|cM| 05:47:32 (None) |
| 07:25.02 | deo | what 2703|cM| ? |
| 07:25.33 | WIMPy | The parameters passed to MeetMe. |
| 07:25.43 | WIMPy | Data=parameters |
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| 07:27.26 | deo | WIMPy: ? so what does it mean exactly? sorry not really familiar with this :( |
| 07:28.07 | WIMPy | You can find the meaning with 'core show application meetme'. |
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| 07:32.15 | schmidts | good morning |
| 07:32.25 | deo | WIMPy: thank you :) |
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| 08:28.05 | apurvtwr | hi everyone. I am working with Asterisk 10 and am trying to redirect an ongoing 2 party call to a conference room using confBridge. To do this, I used redirect Action by passing channel, extrachannel, context, exten, priority, extraContext, extraExten and ExtraPriority |
| 08:28.39 | apurvtwr | with this it hangsup the leg corresponding to extrachannel |
| 08:29.16 | WIMPy | IIRC there was a bug with that effect. Try to upgrade. |
| 08:30.36 | apurvtwr | ok. thanks WIMPy |
| 08:31.05 | apurvtwr | I was looking up some bug reports and found something similar there, turned out the bug as suspended |
| 08:31.13 | apurvtwr | I will look up and upgrade asterisk |
| 08:31.15 | *** join/#asterisk deo (~deo@203.177.214.75) |
| 08:31.48 | WIMPy | Oh, it's not fixed? |
| 08:34.37 | apurvtwr | I can't say for sure. The bug report was not "exactly" the same, but similar. https://issues.asterisk.org/jira/browse/ASTERISK-19191 and https://issues.asterisk.org/jira/browse/ASTERISK-17916 |
| 08:34.39 | LieutPants | [ASTERISK-19191] [Status: Closed] In AMI Redirect action somtimes channels get hangup while redirecting channels to meet me room - https://issues.asterisk.org/jira/browse/ASTERISK-19191 |
| 08:34.39 | LieutPants | [ASTERISK-17916] [Status: Open] First DAHDI line transferred into ConfBridge hung up when second DAHDI line transferred into conference - https://issues.asterisk.org/jira/browse/ASTERISK-17916 |
| 08:35.41 | apurvtwr | they are using redirect action only with meetme |
| 08:36.12 | WIMPy | I don't think the destination matters. |
| 08:37.32 | apurvtwr | yes, I guessed the same. The asterisk logs only show redirecting 1 leg of the call. |
| 08:37.48 | apurvtwr | thanks again. Let me try upgrading it. |
| 08:37.53 | WIMPy | Add your findings to the issue and ask on #asterisk-bugs for the issue to be reopened. |
| 08:38.04 | apurvtwr | hm. sure would. |
| 08:38.12 | WIMPy | Unless it has been fixed, off course. |
| 08:38.19 | apurvtwr | yeh :-) |
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| 08:48.20 | din3sh | hello all |
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| 09:18.36 | ashd | I am trying to get a Bria phone (on osx) to use a hard phone (cisco 7941) connected to asterisk 10.x, i can see the subscribe sip request but asterisk is telling the soft phone "SIP/2.0 401 Unauthorized" i am sending dest-ext:pass@pbx.local as the destination of the subscription … To enable "desk phone" mode in bria, which has a one line SIP uri function. is there any special formatting i need to send. both extensions are working fine if |
| 09:20.20 | ashd | just to clarify, i would like to confirm the syntax of the subscribe uri for use in asterisk. currently dest-ext:pass@pbx.local |
| 09:26.15 | Rhomber | hmm, i was here last night discussing the host= line in a sip.conf peer and that my peer tends to send back on one of two ip's.. i overcame the problem by creating a template and added a peer config for each ip |
| 09:26.22 | Rhomber | this isn't really a good long term option |
| 09:26.42 | Rhomber | my peer is on a /27.. so 31 rules.. and possibly 31 connections / registrations.. not good |
| 09:26.53 | WIMPy | It's the only option. |
| 09:26.54 | *** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no) |
| 09:26.55 | Rhomber | sorry 29 rules |
| 09:27.09 | Rhomber | isn't that a bug then? |
| 09:27.14 | WIMPy | I also have 36 peers for one of my ITSPs. |
| 09:27.26 | Rhomber | lol, at least im not alone |
| 09:27.33 | WIMPy | If it is something, then a missing feature. |
| 09:28.00 | WIMPy | Or you allow guest calls and don't care where they come from. |
| 09:28.13 | Rhomber | i guess, but id call it more of a design flaw |
| 09:28.23 | Rhomber | i.e. it was designed/implemented wrong in the first place |
| 09:28.39 | Rhomber | yeah, i noticed that option too.. not a fan |
| 09:28.43 | WIMPy | Taht's true for a lot of things. |
| 09:28.55 | Rhomber | fair enough :) |
| 09:28.57 | WIMPy | But this one can be worked around easily. |
| 09:29.04 | Rhomber | ok |
| 09:29.36 | Rhomber | beyond config, does it create multiple connections to the peer?.. or is that just how it appears when viewing the sip connections in the manager? |
| 09:30.31 | WIMPy | SIP doesn't have connections at all. |
| 09:31.25 | Rhomber | in "sip show peers" that is |
| 09:31.46 | Rhomber | true, i should have guessed.. just sessions/tokens? |
| 09:31.57 | WIMPy | That jst shows the configured peers. |
| 09:32.13 | Rhomber | and so, the additional peers will only create sessions when using them outbound? |
| 09:32.53 | Rhomber | just trying to get a feel for whether there is any overhead beyond config |
| 09:33.07 | WIMPy | SIP is stateless. So there isn't much that's created beyond the calls. |
| 09:33.31 | WIMPy | Just the list of peers to scan when a call comes in, really. |
| 09:35.02 | Rhomber | ok, that's reasonable then |
| 09:35.10 | Rhomber | as long as they don't go adding any more ranges |
| 09:36.24 | Rhomber | thanks :) |
| 09:37.45 | WIMPy | That's the main issue. You have to keep the list up to date. |
| 09:38.49 | *** join/#asterisk DND (5ec8071a@gateway/web/freenode/ip.94.200.7.26) |
| 09:38.52 | DND | hi guys |
| 09:39.10 | DND | i need to know what is usually the problem when it has blu/red alarm |
| 09:39.57 | WIMPy | Nothing connected. |
| 09:40.04 | *** join/#asterisk areski (~areski@81.184.35.151.dyn.user.ono.com) |
| 09:40.16 | Rhomber | WIMPy: true |
| 09:42.27 | DND | WIMPy: is that for me? |
| 09:42.38 | WIMPy | yes |
| 09:42.47 | DND | so does this means there's no signal from the telco |
| 09:43.00 | WIMPy | yes |
| 09:44.15 | DND | because right now its connected to the telco |
| 09:44.15 | Rhomber | WIMPy: I guess I should have a separate config for inbound then |
| 09:44.32 | Rhomber | since qualify=yes actually does mean it keeps check on the peers |
| 09:44.34 | DND | i did dahdi_scan and gives blu/red |
| 09:44.49 | Rhomber | get a whole bunch of "chan_sip.c:29125 sip_poke_noanswer: Peer 'alltel-1-72' is now UNREACHABLE! Last qualify: 0" |
| 09:45.20 | *** join/#asterisk Dibbler (~Dibbler@host86-150-168-203.range86-150.btcentralplus.com) |
| 09:45.38 | WIMPy | Rhomber: What LEDs do you have? What do they indicate? |
| 09:45.50 | WIMPy | Err. sorry. DND. |
| 09:47.09 | DND | sorry for what? |
| 09:47.38 | WIMPy | Line status |
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| 09:55.30 | DND | WIMPy: is there any way to test if i really cant get any signal from telco? |
| 09:56.33 | *** join/#asterisk vlad_starkov (~vlad_star@194.186.220.230) |
| 09:56.34 | din3sh | how to get Caller ID on a phone where an agent has logged in? |
| 09:56.36 | WIMPy | What LEDs do you have? What do they indicate? |
| 09:57.45 | DND | im using a TE121 from digium and i see the red light blinking |
| 09:58.26 | WIMPy | And on the NT? |
| 09:58.57 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 09:59.12 | DND | nt? |
| 09:59.48 | WIMPy | The thing you (hopefully) plugged the other end of the line in to. |
| 10:03.36 | DND | its plugged in from the other end. im just confirming from my colleague the light status |
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| 10:04.36 | WIMPy | And you should find some LEDs there. |
| 10:07.22 | DND | WIMPy: it seems there's no device/modem |
| 10:07.40 | DND | its directly connected to the telco |
| 10:07.45 | WIMPy | There must be. |
| 10:07.52 | WIMPy | That doesn't work. |
| 10:08.05 | WIMPy | It's the wrong kind of interface. |
| 10:09.48 | DND | here's the other end http://i.imgur.com/5EEZw0h.jpg |
| 10:10.43 | WIMPy | Looks like you're missing an NT. |
| 10:12.36 | Rhomber | hmm, it would seem that this won't work |
| 10:12.49 | Rhomber | i guess i'll have to use freeswitch or something more practical.. |
| 10:13.23 | Rhomber | i need qualify=yes, and the remove end could be any one of those IP's |
| 10:13.33 | Rhomber | so the 'register' won't know which peer config to use |
| 10:13.49 | kaldemar | Rhomber: a register does not use peer configs. |
| 10:14.07 | WIMPy | Register statements are not linked to peers in any way. |
| 10:14.11 | *** join/#asterisk linocisco (~linocisco@203.81.67.114) |
| 10:14.14 | DND | can you confirm what NT means? maybe its called different? |
| 10:14.18 | DND | from here |
| 10:14.23 | Rhomber | well im not sure whats going on then :( |
| 10:14.42 | kaldemar | Rhomber: are you worried about the unreachable status? |
| 10:17.28 | kaldemar | and why would you need qualify=yes on itsp peers? |
| 10:19.45 | linocisco | hi all, there is astricon2013 in malaysia in april 29,30. who is going? |
| 10:20.02 | *** join/#asterisk Faustov (user@gentoo/user/faustov) |
| 10:20.13 | Rhomber | if that's the case then it should be fine, as i can choose the ip for outbound |
| 10:20.13 | WIMPy | DND: Network Terminator. It converts from an U to an S interface. |
| 10:20.13 | Rhomber | mabey i got myself banned from the provider with the first config (where everything was qualify=yes) |
| 10:20.14 | WIMPy | Rhomber: Make one more peer for outbound with the DNS-RR in the host line. |
| 10:20.14 | Rhomber | Registration for '612XXXXXXX@voip.alltel.com.au' timed out, trying again (Attempt #18) |
| 10:20.14 | Rhomber | and so on |
| 10:20.15 | Rhomber | thats what i tried to do with host=voip.alltel.com.au .. got some wierd problems.. as the ip the RR gave me matched one for my inbound definitions |
| 10:20.15 | Rhomber | of course id prefer to have the outbound RR |
| 10:20.15 | Rhomber | but now i can't register at all |
| 10:20.15 | Rhomber | so i dunno |
| 10:20.15 | Rhomber | i think i'll comment it all out and have a look |
| 10:20.32 | *** join/#asterisk chaz68 (~ChuckMast@wsip-24-234-137-89.lv.lv.cox.net) |
| 10:20.59 | Rhomber | yeah, not sure why i get an unreachable.. i might have banned myself |
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| 10:21.11 | *** join/#asterisk FLeiXiuS (~FLeiXiuS@unaffiliated/fleixius) |
| 10:21.14 | Rhomber | and shouldn't i have a qualify statement? |
| 10:21.28 | *** join/#asterisk gerhard7 (~gerhard7@77-172-47-159.ip.telfort.nl) |
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| 10:23.27 | kaldemar | Rhomber: they don't necessarily answer any qualify requests at all. calls may still work. |
| 10:23.49 | Rhomber | the qualify was working fine |
| 10:24.04 | Rhomber | but now register doesn't even work |
| 10:28.56 | *** join/#asterisk DND (5ec8071a@gateway/web/freenode/ip.94.200.7.26) |
| 10:29.16 | DND | hi WIMPy i asked the telco to come to the office and test the line again |
| 10:29.27 | DND | if its really working or now |
| 10:29.30 | DND | *not |
| 10:30.15 | linocisco | Dear all, I would like to have asterisk embedded cheapest settop box for hotel use of 50 FXS line with analog phones with 4 FXO Landline including GSM sim card gateway |
| 10:30.19 | WIMPy | With what equipment? |
| 10:30.23 | linocisco | where can I find? |
| 10:31.31 | Rhomber | I blew away all of my sip config for the peer sections.. register is the only problem |
| 10:31.43 | Rhomber | i may have banned myself.. or they may have blown up |
| 10:31.46 | Rhomber | who's to tell |
| 10:32.20 | DND | i dont know what equipment |
| 10:32.33 | DND | its in jordan. and im here in uae |
| 10:32.43 | DND | but i can confirm that once they come to the office |
| 10:33.09 | linocisco | Dear all, I would like to have asterisk embedded cheapest settop box for hotel use of 50 FXS line with analog phones with 4 FXO Landline including GSM sim card gateway |
| 10:33.23 | *** join/#asterisk apurvtwr (b64722fb@gateway/web/freenode/ip.182.71.34.251) |
| 10:33.50 | WIMPy | DND: How has that even been set up? Did someone cut apart a patch cable and put it in to the punch block?? |
| 10:34.00 | *** join/#asterisk threesome (~threesome@ip-94-113-12-74.net.upcbroadband.cz) |
| 10:34.12 | DND | WIMPy: it seems like it |
| 10:34.33 | DND | i dont know how they do that there. here in dubai we have proper modem |
| 10:35.14 | WIMPy | You definitely need that. |
| 10:36.31 | DND | kinda frustrating. because they keep following up for the payment but its not even setup properly. |
| 10:36.46 | DND | middle easterns.... |
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| 11:22.30 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 11:23.10 | Chainsaw | So, Asterisk 11.2.1 with DAHDI 2.6.2... I seem to have a delay of *exactly* 10 seconds before a SIP call is acted upon. I don't see events with sip debug or rtp debug. |
| 11:23.41 | Chainsaw | Just a random act of desperation, but has anyone seen that kind of behaviour before? It started after a reboot of the box. |
| 11:24.15 | Tim_Toady | dns issues? |
| 11:24.17 | Chainsaw | (After the delay, the call is handled normally and events complete as if nothing ever went wrong. But it's long enough that devices & people operating them give up.) |
| 11:24.32 | *** join/#asterisk sedeki (~textual@unaffiliated/sedeki) |
| 11:24.35 | Chainsaw | Tim_Toady: That's what I thought at first, but it happens in two offices. |
| 11:24.53 | Chainsaw | Tim_Toady: And the phones there use different DNS servers. I'll swap the order on the phone system box all the same though. |
| 11:25.04 | sedeki | hello all. i apt-get:ed asterisk. is hangup.so supposed to be included? |
| 11:25.31 | Tim_Toady | watch the dialplan excecution to see where it stucks, and also a tcpdump of the sip traffic |
| 11:25.45 | *** join/#asterisk davlefouAMD (~david@41.225.34.79) |
| 11:25.57 | sedeki | Tim_Toady how does one view the dialplan execution? |
| 11:26.09 | teff | Chainsaw, had same problem with after I compiled it on my raspberrypi using SIP only trunk, I assumed it was just the rpi not coping but seems much more responsive on the precompiled version of 1.6 from repository |
| 11:26.11 | Chainsaw | Tim_Toady: It does not hit the dialplan until after the 10 seconds. |
| 11:26.11 | sedeki | i'm using pycall right now but there is not error output |
| 11:26.18 | Tim_Toady | connects to the asterisk console and sets verbose to 3 |
| 11:27.05 | teff | but I also had to reimage due to corruption so it might be unrelated |
| 11:27.07 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.182) |
| 11:31.29 | kaldemar | sedeki: such module does not exist. what are you really looking for? |
| 11:31.43 | sedeki | the quick start docs had me add it |
| 11:31.57 | kaldemar | which docs? |
| 11:32.03 | sedeki | app_hangup.so |
| 11:32.12 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:32.20 | sedeki | http://www.asterisk.org/sites/asterisk/files/mce_files/documents/asterisk_quick_start_guide.pdf |
| 11:34.10 | kaldemar | the hangup application is not a even separate module. |
| 11:36.16 | kaldemar | sedeki: besides, you'll be better off just skipping those load lines and use autoload=yes. |
| 11:38.11 | kaldemar | that really is an error in the document. there has never been an app_hangup.so. |
| 11:38.52 | sedeki | how can I see a dialplan execution? for debugging |
| 11:39.27 | kaldemar | attach to CLI with "asterisk -vvvr" when you have asterisk running. |
| 11:39.30 | kaldemar | ~book |
| 11:39.30 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 11:39.48 | kaldemar | you'll find better documentation there. |
| 11:40.55 | *** join/#asterisk marksaitis (~marksaiti@gemsyorkroad.demon.co.uk) |
| 11:44.36 | Greenlight | Asterisk has just deadlocked again, aside from "gdb asterisk --pid=<* pid>" before killing it, what else can I do to give a hint as to why it's deadlocked? |
| 11:47.44 | Greenlight | Hmm... what useful info should I try and grab from gdb ? |
| 11:51.03 | *** join/#asterisk FireAndIce (~FireAndIc@175.100.158.242) |
| 11:52.14 | *** join/#asterisk chris_n (~Chris@koha/developer/chris-n) |
| 11:53.18 | *** join/#asterisk italorossi (~italoross@187.60.66.11) |
| 12:03.07 | kaldemar | Greenlight: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace#GettingaBacktrace-GettingInformationForADeadlock |
| 12:05.27 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 12:07.05 | Greenlight | I got the "thread apply all bt" output, but havn't been runninmg DEBUG_THREADS, so can't get lock information (I don't think). Hopefully the gdb output will give some indication. |
| 12:07.57 | file | throw it up on a pastebin and I'll take a gander right now |
| 12:08.56 | Greenlight | Ok, will do, gimme two mins. Much appricaited! |
| 12:11.28 | Greenlight | Hmm exceeded limit of pastebin.com, know of one with a larger limit ? |
| 12:11.45 | *** join/#asterisk fakhir (~fakhir@unaffiliated/fakhir) |
| 12:12.24 | file | http://paste.ubuntu.com/ perhaps? |
| 12:13.02 | Greenlight | Yup - http://paste.ubuntu.com/5627997/ |
| 12:13.33 | Greenlight | That's an output of "info thread", followed by "thread apply bt all" |
| 12:13.33 | file | huh |
| 12:14.40 | linocisco | infobot, is it 3rd edition book? |
| 12:16.40 | file | something is holding the lock for the channels container |
| 12:16.55 | Greenlight | Yea, that's what I thought from just the CLI |
| 12:16.56 | file | and is just either deadlocked itself or never released it |
| 12:18.13 | file | I _think_ I know what it is... |
| 12:18.25 | file | uno momento |
| 12:18.30 | Greenlight | goes to start ordering file beer |
| 12:20.08 | *** join/#asterisk mahlon (~mahlon@martini.nu) |
| 12:21.30 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 12:22.18 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:23.05 | *** join/#asterisk FireAndIce (~FireAndIc@123.201.2.126) |
| 12:26.45 | *** join/#asterisk vlad_sta_ (~vlad_star@109.188.127.41) |
| 12:27.07 | *** join/#asterisk TimeRider (~steve@timerider.plus.com) |
| 12:29.04 | sedeki | does anyone know what this means? "Unable to create channel of type 'SIP' (cause 20 - Unknown)" |
| 12:30.14 | kaldemar | sedeki: asterisk does not know an ip address for a peer you tried to dial. |
| 12:30.31 | file | hum hum hum |
| 12:31.07 | sedeki | my extension looks like this exten => 2600,1,Dial(SIP/100&SIP/VoIPProvider/*some_number_here,150,r,t,) |
| 12:31.26 | *** join/#asterisk vlad_starkov (~vlad_star@109.188.126.43) |
| 12:31.31 | *** join/#asterisk kesselklopfer79_ (~kesselklo@astaro.starface.de) |
| 12:31.48 | sedeki | why do i need to specify 100&SIP? which is my local incoming user |
| 12:31.51 | kaldemar | sedeki: ,r,t, is not valid btw. |
| 12:32.20 | kaldemar | SIP/100 makes asterisk expect you have [100] in sip.conf and a known address for it. |
| 12:32.31 | kaldemar | "sip show peers" |
| 12:33.35 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 12:33.36 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 12:33.46 | sedeki | I have 100, but the host is unknown. i've treated it as a user login because that's what happens. I also have VoIPProvider/some_number_here with known host |
| 12:34.22 | Greenlight | All of the threads, from what I can see are in either "poll", "pthread_cond_timedwait", or "__lll_lock_wait". They *all* seem to be waiting on that lock being released |
| 12:35.29 | file | I'm past that now. |
| 12:35.36 | Greenlight | :) |
| 12:36.06 | *** join/#asterisk vlad_sta_ (~vlad_star@83.220.239.213) |
| 12:36.09 | file | the device state stuff is attempting to get the lock for a Local channel, but seemingly something else has it |
| 12:36.13 | file | which causes stuff to lock up all over |
| 12:36.56 | kaldemar | sedeki: unknown host causes the warning. the other one is called if asterisk knows how. |
| 12:37.08 | linocisco | hi all, why do we write for dialplan applications? what is dialplan application? I simply see that we can define dialplan as easy in freepbx through webmin. Why is that? |
| 12:37.49 | sedeki | kaldemar what do you mean? I've changed 100&SIP to VoIPProvider&SIP because that's what I want to call through |
| 12:37.52 | [TK]D-Fender | linocisco, ... |
| 12:37.53 | [TK]D-Fender | ~book |
| 12:37.54 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 12:37.54 | [TK]D-Fender | ^^^ |
| 12:38.05 | sedeki | kaldemar what is the other VoIPProvider? "knows how"? |
| 12:38.11 | [TK]D-Fender | linocisco, dialplan = exttensions.conf which is what tells * how to process your calls |
| 12:38.17 | kaldemar | sedeki: changed to what? show the whole line. |
| 12:38.44 | [TK]D-Fender | linosAf for "why" ... its the instructions that say what to DO with the calls. There is no assumed behaviour, it's up to you to tell * how to work |
| 12:38.45 | sedeki | exten => 2600,1,Dial(SIP/VoIPProvider&SIP/VoIPProvider/*some_number_here,150) |
| 12:38.53 | kaldemar | linocisco: you're free to use a GUI and be limited to the functionality that it offers you. |
| 12:39.07 | kaldemar | sedeki: that makes no sense. |
| 12:39.23 | sedeki | kaldemar is the first VoIPProvider where *what* to call should go? |
| 12:39.39 | [TK]D-Fender | linocisco, I can make a dialplan I code myself do ANYTHING I want. FreePBX has some BASIC ideas that it can do... but if you want to get creative then you're in for real trouble |
| 12:39.59 | linocisco | kaldemar, thanks bro. Can I see some sample improved dialplan applications and see how it works then default existing one by system? |
| 12:40.02 | kaldemar | sedeki: no. SIP/VoIPProvider and SIP/VoIPProvider/*some_number_here are completely separate. you're dialing two destinations at the same time. |
| 12:40.23 | kaldemar | linocisco: you're making questions (kind of) that can't be answered. |
| 12:40.26 | [TK]D-Fender | linocisco, What do you mean "improved dialplan applications"? |
| 12:40.31 | sedeki | exten => 2600,1,Dial(SIP/VoIPProvider/*some_number_here,150) <--- is this correct? |
| 12:40.46 | kaldemar | sedeki: by syntax, yes. |
| 12:40.51 | sedeki | let me try it |
| 12:40.51 | Greenlight | file: Is this in thread #488? |
| 12:41.03 | file | Greenlight, that's as much as I can dissect from this - I think the lock happened in the past and isn't going to show up in here... |
| 12:41.11 | [TK]D-Fender | linocisco, Dialplan application are the COMMANDS you use in extensions.conf to process your calls - eg: Dial() Read(), WaitExten(), etc |
| 12:41.12 | file | Greenlight, yes |
| 12:41.19 | linocisco | [TK]D-Fender, sorry for confusion. I have instgalled Freepbx once and found I can create dialplan according to the options system provided. |
| 12:41.33 | kaldemar | sedeki: but only if you have [VoIPProvider] in sip.conf and it has the required settings to dial via that destination. |
| 12:41.35 | [TK]D-Fender | linocisco, You don't create anything.. FreePBX does it for you. |
| 12:41.48 | [TK]D-Fender | linocisco, And it isn't anything you can customise easily |
| 12:41.49 | file | Greenlight, I could probably give you a patch to make it less of an impact at least |
| 12:41.56 | Greenlight | yea :) |
| 12:41.56 | sedeki | kaldemar yes I do |
| 12:42.06 | sedeki | kaldemar i get this error "Auto fallthrough, channel 'SIP/100-00000004' status is 'CONGESTION'" |
| 12:42.08 | Greenlight | Where are you seeing the Local channel bit ? |
| 12:42.08 | linocisco | [TK]D-Fender, but nowadays, most people are using programming langauges for diaplan scriptings. I dont know why |
| 12:42.33 | [TK]D-Fender | linocisco, Because most people only need the very little the GUI's offer and that's enough. People are lazy |
| 12:42.36 | linocisco | [TK]D-Fender, ok i see. |
| 12:42.48 | file | Greenlight, line 3 in thread 488 has obj=0x7f0b388645b8 search the file for that and you' |
| 12:42.53 | file | erm you'll see it's a Local channel |
| 12:43.19 | linocisco | [TK]D-Fender, do I need to learn programming langauges to write dialplan applications? |
| 12:43.36 | kaldemar | sedeki: don't paste single lines here. use pastebin and always use the whole output for a call. enabling sip debug is also often required to see what is wrong. |
| 12:43.47 | [TK]D-Fender | linocisco, You keep talking about "writing dialplan applications"... I don't think you understand what that term means |
| 12:44.01 | linocisco | [TK]D-Fender, that can be true |
| 12:44.01 | Greenlight | Ahh I see. Can device state stuff be disabled ? |
| 12:44.07 | [TK]D-Fender | linocisco, DIAL is an application. Why would you need to write one? You need to explain your goal better |
| 12:44.15 | sedeki | kaldemar ok, hang on |
| 12:44.20 | file | Greenlight, no |
| 12:44.32 | Chainsaw | [TK]D-Fender: I think it's another case of wanting to dial to connect "phone A" to "phone B" with an ampersand. |
| 12:44.51 | [TK]D-Fender | Chainsaw, Don't try to play psychic with this one, it's a nasty moving target |
| 12:45.14 | Chainsaw | powers the crystal ball down and sulks in a corner |
| 12:45.15 | Greenlight | Ok. It seems to be in process of dealing with a bridge (Bridge/ being part of a masqurade?) I think in that thead, would that be correct ? |
| 12:45.32 | linocisco | [TK]D-Fender, ok bro |
| 12:45.37 | linocisco | [TK]D-Fender, ok bro be right back |
| 12:45.45 | file | Greenlight, it's processing a device state change for a Bridge channel |
| 12:46.10 | [TK]D-Fender | BRB as well |
| 12:46.34 | file | does some reading of code |
| 12:46.59 | sedeki | kaldemar http://pastebin.com/aZKbPwpH |
| 12:47.14 | Greenlight | Hmm...why would it then need to get a lock on a Local channel? (Sorry for silly questions - trying my best to understand what's going on, in the likely futile effort to remedy it!) |
| 12:47.30 | file | Greenlight, the state is unknown so it is iterating channels to see if one is present which would mean it is in use |
| 12:47.56 | Greenlight | Ah, so for each channel, it locks it, checks, then should unloc k? |
| 12:48.03 | file | yes |
| 12:48.09 | Greenlight | As it goes over the channel list. Ok gotcha |
| 12:48.28 | Greenlight | But something, somewhere, has left this particular Local channel locked |
| 12:48.33 | kaldemar | sedeki: is there going to be a follow-up for that? |
| 12:48.34 | file | yes |
| 12:48.47 | sedeki | kaldemar how do I enable the log? |
| 12:49.12 | Greenlight | And, this bridging thread is also holding a lock on the list of channels ? |
| 12:49.13 | sedeki | i have core set debug 3 now |
| 12:49.29 | file | the device state thread holds a lock on the list of channels when going through |
| 12:50.46 | Greenlight | But, if we were to patch the device state bit to be a bit more friendly when this happens, would something else which trawls the list of channels, sooner or later not enter the same deadlock? |
| 12:50.57 | *** join/#asterisk [TK]D-Fender (~Joe@216-191-106-165.dedicated.allstream.net) |
| 12:51.18 | file | yeah I can't fix this without either a core show locks or maybe a full log file with debug... I could only buy you some time |
| 12:51.58 | Greenlight | Are there lots of places that lock the list and scan through it ? |
| 12:52.29 | file | a few |
| 12:53.30 | Greenlight | So chances are that once that local channel is left "dead", it's going to go messy fairly quickly |
| 12:53.41 | file | hum... I wonder... |
| 12:53.42 | file | yes |
| 12:55.15 | *** join/#asterisk MrMeek (~meekhime@172-4-223-5.lightspeed.toldoh.sbcglobal.net) |
| 12:55.18 | *** join/#asterisk serafie (~erin@nat/digium/x-msuotwktsulhxwfd) |
| 12:56.00 | Greenlight | As you can see with the 500 threads, it's a rather busy system. Last time I enabled DONT_OPTIMIZE and BETTER_BACKTRACES, the server ground to a halt very very quickly, although this may have been with 1.8 or 10. |
| 12:56.08 | *** join/#asterisk brad_mssw (~brad@shop.monetra.com) |
| 12:56.17 | file | DONT_OPTIMIZE should be fine, the locking stuff would impact a lot |
| 12:56.36 | Greenlight | Perhaps it was the locking stuff then |
| 12:56.50 | Greenlight | Ah yea DEBUG_THREADS |
| 12:56.50 | file | a full debug log might be your best bet |
| 12:57.00 | sedeki | kaldemar sorry for the stupid questions |
| 12:57.27 | file | DONT_OPTIMIZE plus full debug log - that stuff gets written out in a separate thread so it shouldn't impact things terribly much |
| 12:58.21 | file | actually... I have an idea of what is going on |
| 12:58.34 | sedeki | kaldemar strange what you mentioned earlier that …",r,t" is invalid syntax. it was in the docs |
| 12:59.37 | file | nah not that... nevermind |
| 13:00.30 | file | Greenlight, sorry that's all I've got! |
| 13:01.04 | Greenlight | Ok, thanks so much for getting me this far! |
| 13:01.06 | kaldemar | sedeki: in CLI, "core set verbose 10", "sip set debug on" |
| 13:01.17 | Greenlight | I wonder what's killing that Local channel to begin with] |
| 13:02.06 | Greenlight | Right, I'll turn on debug logging then |
| 13:02.32 | kaldemar | sedeki: which doc has ",r,t"? options are not separated by commas, that should be ",rt". |
| 13:02.34 | file | I *think* it's been hung up, that I'm sorta sure of based on the functions being called and where |
| 13:03.09 | Greenlight | There's 3 things I use local channels for. One of them you're going to cringe at. |
| 13:03.26 | Greenlight | First, is whispering sound, so it goes between ChanSpy and Playback |
| 13:03.38 | Greenlight | Second, is originating a call from a ConfBridge |
| 13:03.55 | Greenlight | The third, is to invoke a System call, so that I can move a file via the AMI |
| 13:04.23 | Greenlight | So, the third Orginates a call to the System dialplan app (!) |
| 13:04.36 | file | those "should" be fine |
| 13:06.02 | sedeki | kaldemar http://pastebin.com/2xT4UHw9 |
| 13:06.44 | sedeki | kaldemar ah, never mind. it was from a website (http://agix.com.au/blog/?p=2656) actually, asterisk did not complain but i guess it is wrong |
| 13:08.11 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 13:08.55 | *** join/#asterisk sedeki_ (~textual@unaffiliated/sedeki) |
| 13:09.07 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.239.213) |
| 13:09.38 | sedeki | kaldemar sorry, internet died. |
| 13:09.45 | sedeki | [14:06:44] <sedeki> kaldemar ah, never mind. it was from a website (http://agix.com.au/blog/?p=2656) actually, asterisk did not complain but i guess it is wrong |
| 13:13.42 | kaldemar | sedeki: asterisk did not complain because after options and a comma follows the URL argument. |
| 13:14.52 | kaldemar | sedeki: "SIP/2.0 404 Not Found" <-- the number you dialed is not valid. |
| 13:14.55 | Greenlight | file: In the debug log, should I have everything else as well (notice, warning etc), or just debug? |
| 13:15.14 | file | everything as well |
| 13:15.38 | Greenlight | Hmm... looking at it actually: full => notice,warning,error,debug,verbose,dtmf,fax |
| 13:15.50 | Greenlight | Full should already have everything in |
| 13:15.55 | file | yes |
| 13:16.07 | Greenlight | But I don't recall seeing debug messages in there |
| 13:17.00 | jmetro | whats the easiest way to set voicemail volgain for one user |
| 13:17.29 | [TK]D-Fender | jmetro, on the box itself |
| 13:18.07 | [TK]D-Fender | jmetro, those options can be global or box-specific. |
| 13:18.08 | file | Greenlight, should be in there |
| 13:18.17 | sedeki | kaldemar do I need to restart asterisk upon changing any conf file? |
| 13:18.21 | Greenlight | It's odd, mostly all verbose messages, no DEBUG |
| 13:18.35 | Greenlight | Unless, there's not many of them? BUt I'd expect more than VERBOSE |
| 13:19.31 | kaldemar | sedeki: generally no. which config file do you mean now? |
| 13:19.34 | file | Greenlight, huh |
| 13:19.56 | file | Greenlight, do "core set debug 9" |
| 13:20.07 | jmetro | [TK]D-Fender Ah..thank you. I forgot about box-specific |
| 13:20.12 | Greenlight | Oh, I need to enable it from there too |
| 13:20.21 | file | apparently. |
| 13:20.29 | file | I wanna say someone introduced a regression |
| 13:20.46 | sedeki | kaldemar sip.conf and extensions.conf |
| 13:20.50 | Greenlight | maybe when they made each CLI have sepraate verbosity levels |
| 13:20.59 | sedeki | kaldemar btw, what does the star signify at the beginning of a number? |
| 13:21.00 | file | yeahhhhh that's my feeling |
| 13:21.57 | *** join/#asterisk sekil (~sekil@78.24.104.73) |
| 13:22.05 | Greenlight | Hmm yea, that does the trick. Hmm... gonna stick this on the 4TB array, guess it could grow quite quckly |
| 13:22.18 | file | yes |
| 13:22.30 | file | hopefully the issue will appear rapidly |
| 13:22.51 | Greenlight | First was 15 days ago. Then one yesterday, then 3rd today. |
| 13:24.50 | kaldemar | sedeki: a star as in a character. where did you come up with that? |
| 13:25.22 | [TK]D-Fender | <sedeki> kaldemar btw, what does the star signify at the beginning of a number? <- whatever you want it to be |
| 13:25.45 | [TK]D-Fender | sedeki, A number is a number ... you do whatever you want with them |
| 13:25.55 | kaldemar | sedeki: oh, the blog again. don't send it unless you know you need to send such things. |
| 13:27.15 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.239.213) |
| 13:29.03 | sedeki | yeah |
| 13:29.26 | sedeki | got it working but it doesn't receive audio |
| 13:29.45 | Greenlight | 5Mb so far, only had it on a minute. Going to need that disk space ^^ |
| 13:30.32 | kaldemar | sedeki: the "insecure=very" option is also invalid in the configs. is should be "insecure=port,invite" if you use 1.6.X or newer. |
| 13:30.44 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 13:32.42 | kaldemar | sedeki: as for the audio, time for another sip debug. |
| 13:33.00 | sedeki | kaldemar what does port,invite do? |
| 13:33.06 | sedeki | okay, let me output the log. |
| 13:34.10 | skrusty | does anyone know much about using ricoh fax machines with asterisk? They support SIP, can i can get it to make calls out, but it's failing on t.38 |
| 13:35.09 | coppice | the ricohs work OK with FreeSwitch using T.38, so they are probably OK with asterisk too |
| 13:35.31 | coppice | they are one of the few ranges of machines that support fast T.38 over TCP |
| 13:36.04 | *** join/#asterisk quenenni (~quenenni@unaffiliated/quenenni) |
| 13:36.13 | *** part/#asterisk quenenni (~quenenni@unaffiliated/quenenni) |
| 13:36.35 | sedeki | kaldemar http://pastebin.com/PM5M6RJD |
| 13:36.38 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.239.213) |
| 13:36.59 | *** join/#asterisk lorsungcu (~anonymous@65.103.31.33) |
| 13:38.21 | sedeki | how do I route the incoming called number to outgoing? i've seen stuff like "exten => s,1,SIP(……..)" |
| 13:38.27 | sedeki | where s is the called number |
| 13:38.47 | skrusty | ${EXTEN} ? |
| 13:39.02 | *** join/#asterisk jeffspeff (~jeffspeff@12.49.160.131) |
| 13:39.40 | kaldemar | sedeki: it does not require authentication for calls that come from the ip address that the peer has defined. use a sample config to see what options do: http://svn.digium.com/svn/asterisk/tags/11.2.1/configs/sip.conf.sample |
| 13:39.47 | bulkorok | isn't there an application that shows all current channel variables in CLI!? |
| 13:39.48 | sedeki | something like exten => 2600,1,Dial(SIP/VoIPProvider/{EXTEN},150) ? |
| 13:39.51 | sedeki | uh |
| 13:39.59 | sedeki | exten => s,1,Dial(SIP/VoIPProvider/{EXTEN},150) ? |
| 13:40.16 | skrusty | sedeki: yeah |
| 13:40.26 | jeffspeff | is there a way to use res_calendar_ews to use a single account to check the status of users based on email address? |
| 13:40.40 | kaldemar | sedeki: s is used when there is no number or when you request them to send calls to s like you do on the register line with "/s". |
| 13:41.26 | kaldemar | sedeki: "{EXTEN}" is invalid, variables need a $, like ${EXTEN}. however, your in example ${EXTEN} would be "s". |
| 13:42.00 | kaldemar | bulkorok: DumpChan |
| 13:42.11 | bulkorok | ah... I'll check that :-) |
| 13:42.21 | sedeki | I have exten => s,1,Dial(SIP/VoIPProvider/s,150) . it doesn't find the extension |
| 13:42.56 | jeffspeff | you can't dial s,exten |
| 13:42.57 | kaldemar | sedeki: "find the extension" does not mean anything. asterisk does what you tell it to. s is just "s". it does not contain anything. |
| 13:43.24 | jeffspeff | sedeki, you're confusing the Dial() and Goto() syntax |
| 13:44.01 | sedeki | how do I pickup at whatever extension they called? |
| 13:44.12 | sedeki | and then dial my VoIP provider with that extension? |
| 13:44.13 | jeffspeff | ~book |
| 13:44.13 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 13:44.19 | kaldemar | sedeki: using a pattern. |
| 13:44.22 | jeffspeff | please read that |
| 13:44.44 | kaldemar | sedeki: https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching |
| 13:44.48 | *** join/#asterisk leedm777 (~leedm777@nat/digium/x-cfkqwlmxmryarohy) |
| 13:44.50 | *** join/#asterisk Sorcier_FXK (~nssystem@unaffiliated/sorcierfxk) |
| 13:47.28 | *** join/#asterisk ghostmediapro (~Trupsalms@unaffiliated/ghostmediapro) |
| 13:48.50 | ghostmediapro | Debug-in asterisk sip uri from sever2 to server1, asterisk cli logs from both servers server2: http://pastebin.com/NL2qKJnk and server1: http://pastebin.com/nj3urJyp |
| 13:50.52 | sedeki | kaldemar haha I'm matching _! and it calls me twice when I call 123 :) |
| 13:52.21 | kaldemar | sedeki: sure, it matches the hangup extension (h). that's one of the reasons the use of patterns like _! and _. and is discouraged. |
| 13:53.11 | sedeki | It calls me twice when I only have one extension, the one that dials |
| 13:54.30 | kaldemar | use _X. or _X! if you want to match all numbers. |
| 13:55.03 | [TK]D-Fender | ghostmediapro, you left your CALLERID on your first server as the SAME as one that matech on your second. Server 2 thinks that is "702" on IT'S side. Because you are calling without auth, it fails |
| 13:55.35 | [TK]D-Fender | ghostmediapro, Fix your callerID on your trunk-out on server 1 or fix all of your peers to restrict the IP's that can match it. |
| 13:59.24 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 13:59.24 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 14:00.32 | sedeki | kaldemar got it working, thanks for the hint. any idea on how I can receive audio? codec problem? |
| 14:00.54 | *** join/#asterisk gusto (~gusto@ppp-93-104-81-201.dynamic.mnet-online.de) |
| 14:01.58 | *** join/#asterisk CunningPike (~CunningPi@d28-23-24-84.dim.wideopenwest.com) |
| 14:01.59 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 14:02.55 | Katty | GOOD MORNING CUPCAKES. |
| 14:03.50 | jmetro | pan* |
| 14:05.00 | kaldemar | sedeki: are both ends silent? |
| 14:05.07 | sedeki | no just one |
| 14:05.20 | sedeki | i can hear from my telephone but not from my computer |
| 14:05.23 | Katty | sounds like someone is getting the cold shoulder. |
| 14:06.03 | ghostmediapro | [TK]D-Fender: ii understand what your saying about the caller id just a bit, but if i were to change the caller id, wouldn't that mean i would have to actually change all the exten |
| 14:06.57 | kaldemar | sedeki: see with rtp debug that asterisk is receiving rtp from the provider and sending it to your zoiper. |
| 14:07.28 | ghostmediapro | [TK]D-Fender: on server one i have set up a inbound route to allow 74296, either using freenum 74296*1483 or direct 74296 is the out come i was hoping to reach. |
| 14:07.29 | sedeki | does it have to do with "allow" "disallow" conf options do you think? |
| 14:07.41 | *** join/#asterisk HmdP_Mobile (~HmdP_Mobi@D9799130.cm-3-2c.dynamic.ziggo.nl) |
| 14:07.54 | kaldemar | sedeki: configuring codecs for [100] would be good. with disallow and allow. |
| 14:08.33 | sedeki | kaldemar so it doesn't go in VoIPProvider? |
| 14:11.55 | kaldemar | sedeki: why wouldn't it? |
| 14:12.16 | sedeki | does it go in both [100] and [VoIPProvider] or just one? |
| 14:12.34 | kaldemar | codecs are not something you configure for one peer only. they are settings that affect every peer, just like any other parameters. |
| 14:13.34 | ghostmediapro | [TK]D-Fender: Oh Great Guru Thank You, But That Kinda Posses A Problem If Some Is Calling Me With Potentially The Same Setup Of Exten right, and if so how can i fix it on server one's end |
| 14:13.55 | [TK]D-Fender | ghostmediapro, I already told you |
| 14:14.07 | [TK]D-Fender | ghostmediapro, Fix the callerid on the CALLING SIDE |
| 14:14.22 | [TK]D-Fender | ghostmediapro, Or restrict the IP's for your peers on the RECEIVING side so they cannot match |
| 14:17.14 | ghostmediapro | [TK]D-Fender: you did, i'm sorry i didn't under stand the later: Or restrict the IP's for your peers on the RECEIVING side so they cannot match, i did for the sake of testing Fix the callerid on the CALLING SIDE, can you please explain just a little more as to what you mean by restrict the IP's for your peers on the RECEIVING side so they cannot match |
| 14:17.56 | [TK]D-Fender | ghostmediapro, What don't you understand? PERMIT/DENY <- Stop the general public from being able to match that peer |
| 14:18.41 | *** join/#asterisk Leddy (leddy@krypton.evosurge.com) |
| 14:18.45 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 14:18.48 | *** join/#asterisk vlad_starkov (~vlad_star@83.220.239.213) |
| 14:21.03 | ghostmediapro | [TK]D-Fender: server one is no nat, public ip, by firewall, and server two is nat, private ip, one server one allow/deny is both 0.0.0.0 and on server two deny is 0.0.0.0 and allow is 192.168.0.0 |
| 14:22.04 | [TK]D-Fender | ghostmediapro, Go for the first fix.... |
| 14:22.54 | *** join/#asterisk ujjain (ujjain@unaffiliated/ujjain) |
| 14:24.02 | ghostmediapro | again correct me if i'm wrong your first fix was with the caller id, the only way to change the caller id is complete change the exten so i did, i made the new exten 750, and now the calls to go through, thank you for that one, but i want to make sure that if someone else has the same potential set up that they would still be able to call in |
| 14:24.48 | [TK]D-Fender | If it can hit a peer it will block anonymous calls. |
| 14:24.57 | [TK]D-Fender | And auth will fail |
| 14:32.32 | ghostmediapro | safe to assume that show never happen, as other setup will be differnt correct |
| 14:33.59 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 14:33.59 | *** mode/#asterisk [+o putnopvut] by ChanServ |
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| 14:52.22 | *** join/#asterisk thecardsmith_ (~quassel@pdpc/supporter/student/thecardsmith) |
| 14:59.15 | kesselklopfer79_ | is it the right way to patch res_agi in function static enum agi_result launch_asyncagi(struct ast_channel *chan, char *argv[], int *efd) in line setup_env(chan, "async", fds[1], 0, 0, NULL); to get the arguments of AGI(agi:async,arg1,arg2.....) ? |
| 15:00.08 | igcewieling | kesselklopfer79_: you might want to ask on #asterisk-dev |
| 15:00.38 | kesselklopfer79_ | igcewieling: ok thx |
| 15:00.52 | *** join/#asterisk CQ (~chatzilla@89.204.135.49) |
| 15:01.48 | CQ | hello, can someone point me to a howto on how to use asterisk to allow me to call a number at my home, and then dial internationally from there as I would with a callind prefix service? |
| 15:02.17 | CQ | or even what keywords to search for would be helpful |
| 15:03.40 | *** join/#asterisk thecardsmith (~quassel@pdpc/supporter/student/thecardsmith) |
| 15:04.44 | *** join/#asterisk vinhdizzo (~vinh@cpe-66-74-182-202.socal.res.rr.com) |
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| 15:06.01 | CQ | I'm looking at this http://www.voip-info.org/wiki/view/Asterisk+call+forwarding |
| 15:06.25 | CQ | I would like to dial number+pin+forwarding number and have asterisk connect me there |
| 15:07.16 | leifmadsen | pretty straight forward with some dialplan |
| 15:07.23 | igcewieling | CQ: you would accept the call in your dialplan, read the PIN, the read the digits to dial, then dial. |
| 15:07.27 | leifmadsen | Read(), Dial(), and DB() |
| 15:09.31 | CQ | ok, so its possible and its easy... I've never used asterisk before, but it seems operatorone.eu doesn't have a dialin in germany anymore, and I want to make international calls from my cell phone |
| 15:09.39 | sruffell | Or on a good PIN dump the call into https://wiki.asterisk.org/wiki/display/AST/Application_DISA ? |
| 15:10.31 | CQ | thanks, I'll keep this info and give it a shot over easter probably. |
| 15:12.24 | *** join/#asterisk freeedrich|_ (friedrich@perplexa.be) |
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| 15:14.39 | Katty | my ticketing software is down |
| 15:14.40 | Katty | I"M LOST |
| 15:15.03 | *** join/#asterisk sruffell (~sruffell@asterisk/the-kernel-guy/sruffell) |
| 15:15.03 | drmessano | Did you open a ticket? |
| 15:15.03 | *** mode/#asterisk [+o sruffell] by ChanServ |
| 15:15.31 | Katty | i totally did. |
| 15:15.32 | Katty | in notepad. |
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| 15:40.16 | igcewieling | I love Adtrans, but sometimes they really piss me off. |
| 15:40.45 | lorsungcu | tell me more about why they piss you off. |
| 15:40.47 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 15:41.52 | jeev | any ringcentral users for faxes here? |
| 15:45.50 | *** join/#asterisk Rhomber (~Rhomber@60-240-245-17.static.tpgi.com.au) |
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| 15:46.51 | Rhomber | would anyone have any ideas why i hear a cracking/jitter noise on csipsimple (asterisk).. all other sip devices work fine.. and also when i leave a message on the asterisk server from csipsimple it sounds crystal clear.. |
| 15:47.37 | Rhomber | i've gone through all the csipsimple settings and nothing seems to help.. is there something i can do in asterisk? |
| 15:49.10 | Rhomber | i just disabled all codecs except PCMU 8Khz, still no love |
| 15:49.28 | Qwell | If it's jitter, then it's your network. |
| 15:49.56 | Rhomber | how does that make sense given what i said above? |
| 15:49.57 | *** join/#asterisk blee (~blee@68.204.217.123) |
| 15:50.18 | Qwell | You described two separate audio streams, in two different directions. |
| 15:50.21 | Rhomber | hmm, with the exception this is via WiFI |
| 15:50.27 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 15:50.35 | Rhomber | not the direction.. i have other SIP devices working fine i mean |
| 15:50.42 | Rhomber | (both directions) |
| 15:50.51 | Chainsaw | WiFi a QoS-free nightmare? Packet loss and jitter through the roof? You don't say?! |
| 15:50.53 | lorsungcu | so the one device on wifi is giving you audio problems |
| 15:50.55 | Rhomber | but yeah, the android phone is connected via WiFi |
| 15:51.20 | lorsungcu | i am stumped, rhomber. |
| 15:51.35 | Chainsaw | lorsungcu: Yeah, it doesn't make sense to me either. Let's get back to our beer. |
| 15:51.45 | Qwell | wait, we were supposed to bring beer? |
| 15:51.47 | Qwell | stops hiding his |
| 15:51.55 | Rhomber | ok, i'll switch my laptop to wifi and do a speed test |
| 15:52.10 | Qwell | speed test isn't going to tell you anything whatsoever |
| 15:52.11 | lorsungcu | speed test = voip test |
| 15:52.13 | *** join/#asterisk santa0536 (~santa@cn-bgp-nat.portaone.com) |
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| 16:07.41 | *** join/#asterisk Rhomber (~Rhomber@60-240-245-17.static.tpgi.com.au) |
| 16:08.09 | Rhomber | while on wifi the laptop performed exactly the same.. sip calls were equally as good and speedtest.net was good |
| 16:08.30 | Rhomber | with the android phone 1m away from the access point.. it still had jitter |
| 16:09.01 | lorsungcu | but you have not connected an ethernet cable to your cell phone and tried then? |
| 16:09.12 | Rhomber | seemed to be simmilar with the 3 softphones.. 3XC and Sipdroid too |
| 16:09.24 | Rhomber | lol, i would if i could |
| 16:09.36 | Rhomber | can i buy an android usb to ethernet? |
| 16:10.00 | Rhomber | oh it seems i can |
| 16:10.04 | Rhomber | but yeah, i don't have one |
| 16:10.23 | lorsungcu | sounds like youre fucked |
| 16:10.26 | lorsungcu | :/ |
| 16:13.12 | *** join/#asterisk _Corey_ (~corey@173-161-229-46-Philadelphia.hfc.comcastbusiness.net) |
| 16:13.47 | *** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey) |
| 16:13.54 | Rhomber | so, solution would be to get a better Wifi AP? |
| 16:15.42 | Rhomber | though im not sure the WRT54G would be the problem |
| 16:15.55 | pabelanger | How do you guys trace down high load on your asterisk boxes? |
| 16:17.39 | *** join/#asterisk eXcAliBuR (~eXcAliBuR@206.162.174.6) |
| 16:17.41 | eXcAliBuR | is back |
| 16:17.52 | eXcAliBuR | waits for the warm welcome feeling to wash over him |
| 16:17.53 | *** join/#asterisk timahvo1 (~rogue@41.212.120.45) |
| 16:18.24 | Greenlight | pabelanger: What sort of metric are you seeing the high load on? |
| 16:18.36 | Greenlight | eg, the load average, or CPU% ? |
| 16:19.09 | igcewieling | pabelanger: "top" |
| 16:19.48 | *** join/#asterisk Neptu (~Hej@213-67-244-97-no100.tbcn.telia.com) |
| 16:22.34 | pabelanger | load |
| 16:22.35 | Rhomber | hmm, the call isn't lagged at all.. it's just .. every say 500ms.. it pops/crackles.. actually hurts my ear a bit.. like a high/low freq that isn't audible |
| 16:22.37 | *** join/#asterisk navaismo (~navaismo@189.241.122.125) |
| 16:22.42 | pabelanger | cpu for my asterisk process is low |
| 16:22.45 | pabelanger | 30% |
| 16:23.14 | pabelanger | however, once in a while it spikes to 300% or some crazy number |
| 16:23.27 | pabelanger | but for the most of the day, 30-40% |
| 16:26.14 | eXcAliBuR | is there a way to distinguish between the line keys and speed dial keys? |
| 16:26.20 | eXcAliBuR | for the digium d50 |
| 16:26.54 | *** join/#asterisk cmendes0101 (~cmendes01@wtnl.corp.tierra.net) |
| 16:27.40 | Greenlight | pabelanger: What's being done on Asterisk at the time ? |
| 16:28.07 | *** join/#asterisk Neptu (~Hej@213-67-244-97-no100.tbcn.telia.com) |
| 16:28.11 | pabelanger | Greenlight, mostly app_queue |
| 16:28.34 | pabelanger | 55 queue members logged in |
| 16:28.36 | pabelanger | all SIP |
| 16:28.40 | pabelanger | codec ulaw |
| 16:28.41 | Greenlight | So, not like it's suddenly dealing with an influx or calls ? |
| 16:28.43 | pabelanger | no transcoding |
| 16:29.13 | pabelanger | Yes, I do notice the load increase when more calls are coming in, that is to be expected |
| 16:29.53 | pabelanger | but, cannot think of anything specific in asterisk that is causing the high load |
| 16:30.04 | pabelanger | I suspect asterisk is waiting for something |
| 16:30.12 | Greenlight | Yea, that 300% spike in CPU does seem odd. How long does the spike last for ? |
| 16:30.15 | pabelanger | and trying to trace down the I/O bottleneck |
| 16:30.24 | pabelanger | less then a second |
| 16:31.01 | Greenlight | I know load average includes IO wait, wasn't sure if CPU% did though - if it is IO related though that could explain it |
| 16:32.07 | igcewieling | pabelanger: what version of Asterisk? There have been bugs i the past which cause a similar issue. |
| 16:32.19 | Greenlight | I've not noticed spikes that extremem in CPU usage on either * 10 or 11 and I've had a "top" running almost constantly for months on one screen now. I used queues extensively. |
| 16:34.54 | Greenlight | It's not something odd like powersaving features enabled on the box, something like that ? |
| 16:35.08 | eXcAliBuR | how might i configure a d50 using dpma to make my line key go to a analog line? |
| 16:35.40 | *** join/#asterisk generalhan (~generalha@about/windows/staff/generalhan) |
| 16:37.31 | Rhomber | ooh, Linaphone seems to actually work |
| 16:38.08 | Rhomber | possibly because it's older than the other ones.. dunno.. it also sucks in other ways though |
| 16:38.44 | pabelanger | Greenlight, yes, this is my thought too |
| 16:38.48 | pabelanger | igcewieling, 1.8.7.1 |
| 16:39.00 | pabelanger | which bug are you referring too? |
| 16:42.32 | igcewieling | pabelanger: upgrade |
| 16:42.44 | *** join/#asterisk lanning (~lanning@50-193-22-25-static.hfc.comcastbusiness.net) |
| 16:42.49 | pabelanger | igcewieling, why? |
| 16:42.53 | *** join/#asterisk jzaw (~jzaw@loki.dzki.co.uk) |
| 16:43.26 | igcewieling | because you are running an ancient version of 1.8, at least run a reasonably recent 1.8 |
| 16:43.46 | igcewieling | there have been various bugs but a quick google search shows at least https://issues.asterisk.org/jira/browse/ASTERISK-17196 |
| 16:43.47 | LieutPants | [ASTERISK-17196] [Status: Closed] High CPU Usage / Choppy Sound (res_timing_pthread) - https://issues.asterisk.org/jira/browse/ASTERISK-17196 |
| 16:44.25 | *** join/#asterisk ketas (ketas@ketas6-sixxs.si.pri.ee) |
| 16:44.48 | pabelanger | igcewieling, I am not using pthreads |
| 16:45.20 | igcewieling | pabelanger: take my advice or don't, doesn't matter to me. |
| 16:45.35 | pabelanger | upgrading to upgrade is not a solution. I understand running the latest and greatest is fine. But, I'd rather find the specific issue backport it first and see the result |
| 16:46.02 | igcewieling | pabelanger: then you have a long frustrating road ahead of you with little support from the community for such an old version |
| 16:46.19 | Rhomber | anyone know any projects based on Linphone? |
| 16:46.34 | igcewieling | If you can't upgrade from 1.8.7.1 to a later version of 1.8, then I think you have a policy or design issue. |
| 16:47.37 | pabelanger | correct |
| 16:47.42 | navaismo | Rhomber, project based on linphone? like...? |
| 16:47.49 | pabelanger | I choose to run 1.8.7.1 right backports for stability |
| 16:47.55 | eXcAliBuR | command to see asterisk version? |
| 16:48.03 | Greenlight | core show version |
| 16:53.14 | Rhomber | actually i quite like Linaphone, I just called myself and conferenced myself |
| 16:53.17 | Rhomber | it was awesome |
| 16:53.22 | Rhomber | like a self echo-generator |
| 16:53.28 | Rhomber | more so the multi call support |
| 16:53.39 | Rhomber | umm like an android project based on linphone |
| 16:53.47 | Rhomber | linphone is opensource |
| 16:54.10 | navaismo | dont you say |
| 16:55.27 | Rhomber | say what? :) |
| 17:00.53 | *** join/#asterisk malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd) |
| 17:00.53 | *** mode/#asterisk [+o malcolmd] by ChanServ |
| 17:05.37 | jeffspeff | is there a way to use res_calendar_ews to use a single account to check the status of users based on email address? |
| 17:06.53 | navaismo | Rhomber, so far i heard people want to use linphone with a raspberry pi to create a video-sip-doorbell |
| 17:07.03 | jeffspeff | test |
| 17:07.07 | jeffspeff | s/test/test1 |
| 17:07.14 | jeffspeff | s/test1/test |
| 17:09.12 | Rhomber | that sounds awesome :) |
| 17:09.29 | Rhomber | but they should use a Odroid-U2 |
| 17:09.31 | Rhomber | ;) |
| 17:11.53 | navaismo | googling Odroid-U2 |
| 17:12.21 | Rhomber | http://www.hardkernel.com/renewal_2011/products/prdt_info.php?g_code=G135341370451 |
| 17:12.35 | Rhomber | 2(ish)Ghz Quadcore, 2Gb ram |
| 17:12.43 | Rhomber | 5cm square |
| 17:12.44 | navaismo | too expensive againt the super-mega-awesome Pi |
| 17:12.52 | Rhomber | aw |
| 17:12.58 | navaismo | s/againt/against/ |
| 17:13.05 | Rhomber | for a doorbell, i guess |
| 17:13.32 | Rhomber | lol a bot that knows perl |
| 17:13.33 | Greenlight | Pi is really an awesome bit of kit, especially for the price. |
| 17:13.33 | Rhomber | nice |
| 17:13.48 | *** join/#asterisk marksaitis (~marksaiti@cpc4-rdng22-2-0-cust932.15-3.cable.virginmedia.com) |
| 17:13.49 | Rhomber | i got a Odroid U2 for an XBMC media center |
| 17:14.03 | Rhomber | sadly, im still waiting for the XBMC crew to finish the release needed to make it work |
| 17:14.20 | Rhomber | shouldn't be too much longer |
| 17:15.19 | navaismo | in the Pi works great the Raspbmc |
| 17:15.53 | Greenlight | Yea, mine plays 1080p movies with DTS perfectly. Very impressive! |
| 17:16.26 | navaismo | i only have a complaint the pi cant handle a 1TB usb hard drive |
| 17:16.43 | Greenlight | I just map my local NAS using samba |
| 17:18.06 | Greenlight | EVen the fact you can browse your movie collection with images and synopsis from iPhone is epically cool |
| 17:19.25 | navaismo | right right |
| 17:20.04 | Greenlight | Curse the money I've spent on inferior media players over the years! |
| 17:20.11 | Rhomber | yeah, XBMC handles samba shares natively |
| 17:20.16 | Rhomber | that's all i do |
| 17:20.26 | Rhomber | though im still using XBMC on an actual Xbox generation 1 :) |
| 17:20.33 | Rhomber | and my files are elsewhere |
| 17:20.57 | Rhomber | and yeah, i love XBMC :) |
| 17:21.07 | Rhomber | which reminds me |
| 17:21.20 | Rhomber | i was going to see if there was an android remote control (app) for it |
| 17:21.23 | Greenlight | Get the pi one, you can control it via most TV remotes - helps with the WAF |
| 17:24.37 | navaismo | if my windows phone has one app for that you will find many for android |
| 17:25.01 | Rhomber | cool cool |
| 17:25.08 | Rhomber | and i have an old school xbmc remote |
| 17:25.11 | Rhomber | works fine |
| 17:25.23 | Rhomber | but, https://code.google.com/p/android-xbmcremote/ |
| 17:25.26 | Rhomber | looks cool |
| 17:25.35 | Rhomber | not sure it will work with my old xbmc though |
| 17:25.58 | Rhomber | oh, they are here |
| 17:26.08 | Rhomber | *joins #xbmc-android* lol |
| 17:26.36 | kesselklopfer79_ | :) |
| 17:26.41 | Rhomber | "On incoming call, display who's calling on TV screen and pause video until call is over" |
| 17:26.43 | Rhomber | haha, whoa |
| 17:27.03 | drmessano | * DO not use if teenagers are present in household |
| 17:27.19 | eirirs | Rhomber: "mom is calling, take the phone ASAP" |
| 17:27.45 | drmessano | Wife is calling. RUN |
| 17:27.55 | eirirs | CANCEL |
| 17:28.00 | kesselklopfer79_ | :) |
| 17:28.10 | Greenlight | DIVERT |
| 17:28.19 | Rhomber | LOL |
| 17:28.23 | eirirs | haha |
| 17:28.23 | igcewieling | the solution to the wife calling problem is to not get married. Quite simple, really. |
| 17:28.39 | Rhomber | LOL, logical :) |
| 17:28.39 | Greenlight | Damn, I missed a trick on that one ^^ |
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| 17:29.05 | navaismo | me too |
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| 17:30.06 | drmessano | The trick to marriage is knowing when to get divorced |
| 17:30.27 | drmessano | Jumping out of a plane isn't bad.. Not pulling the handle on the parachute IS |
| 17:32.45 | Rhomber | LOL |
| 17:32.55 | Rhomber | it's not the fall.. but the ground that kills ya |
| 17:41.33 | BludSuckingFiend | or a tree |
| 17:41.41 | BludSuckingFiend | or if you're really unlucky, the space needle |
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| 17:45.13 | igcewieling | drmessano: So the trick to being a serial killer is to know when to stop? |
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| 17:52.42 | drmessano | igcewieling: Yep |
| 17:53.10 | igcewieling | why get married in the first place then? |
| 17:53.32 | carrar | makes the parents happy |
| 17:53.55 | navaismo | stupidity |
| 17:54.03 | navaismo | o wait in my case a kid hehe |
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| 17:55.12 | Rhomber | i think i know why linphone is better.. it has a c backend |
| 17:55.16 | Rhomber | as opposed to java |
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| 17:56.27 | carrar | does linphone allow you to edit the useragent header? |
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| 18:00.27 | Rhomber | i doubt it, it's less configurable than the others |
| 18:00.35 | Rhomber | it tries to do things itself |
| 18:00.53 | Rhomber | i'm downloading the git now to see how it works |
| 18:01.03 | Rhomber | i'm hoping it's a java api to a c library |
| 18:02.12 | Rhomber | and yes, yes it is |
| 18:02.13 | Rhomber | :) |
| 18:02.32 | Rhomber | so if you know java. you probably can add the feature.. haha |
| 18:02.39 | Rhomber | i know more java than c! |
| 18:05.14 | Rhomber | sadly they use eclipse .. not intellij.. evil.. they should just use maven so we can both be happy |
| 18:06.00 | Rhomber | WTF, open the project and AVG complains it's an "XML bomb".. and i was typing at the time.. so it's quarantined a project file |
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| 18:06.45 | drmessano | WPP |
| 18:06.49 | drmessano | Windows People Problems |
| 18:10.34 | Rhomber | yeh. :( |
| 18:10.38 | Rhomber | oh well, solved it |
| 18:10.47 | Rhomber | will have a play with the source code some other day |
| 18:10.49 | Rhomber | cya all |
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| 18:32.49 | jeffspeff | is there a way to use res_calendar_ews to use a single account to check the status of users based on email address? |
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| 18:58.55 | eXcAliBuR | >:( |
| 19:02.50 | navaismo | :P |
| 19:03.59 | eXcAliBuR | someone has been eatting my cookies |
| 19:04.09 | eXcAliBuR | i keep a box in my desk |
| 19:06.47 | navaismo | Oh man you just dont levae a box of cookies alone |
| 19:12.45 | jmetro | i have a box of cookies in my car that SOMEONE keeps eating. |
| 19:13.01 | jmetro | that someone being me, because they are samoas and delicious. |
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| 19:34.01 | igcewieling | I learned to make cookies so I'd not go bankrupt buying them. 8-) |
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| 19:42.52 | [sr] | hi WIMPy |
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| 21:33.01 | AlexForster | AMI question: I seem to remember reading that only a single Action can be outstanding on a connection - was that correct, and does that still apply to 1.6? 1.8+? |
| 21:33.18 | Qwell | define "outstanding" |
| 21:34.17 | AlexForster | (send Action #1) --time--> (send Action #2) --time--> (receive Response #1) --time--> (receive Response #2) |
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| 21:34.56 | AlexForster | just need to know if, in a multithreaded app, i need to issue actions serially |
| 21:35.11 | WIMPy | no |
| 21:36.28 | AlexForster | cool, ActionID would be pretty pointless if that were the case, but I swear I read it somewhere...probably on voip-info, which I should know by now is always incorrect |
| 21:37.18 | WIMPy | I don't remember that it has been neccessary to wait. |
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| 21:40.05 | AlexForster | another question I just remembered- is Action: Ping ever necessary on TCP? |
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| 21:40.42 | WIMPy | Meccessary for what? |
| 21:41.58 | AlexForster | I guess for anything, but I was thinking specifically to keep the session alive |
| 21:42.40 | WIMPy | Only if you have a really bad router in between. |
| 21:43.17 | AlexForster | hmm |
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| 21:52.42 | edong23 | <Rhomber> i guess i'll have to use freeswitch or something more practical.. |
| 21:53.02 | edong23 | whats the issue again? |
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| 22:06.11 | DarthExpeditor | Ok. I got an odd one. I'm working on some code with an arduino and I am trying to add a char to theend of a String and its giving me some trouble. At first I tried myReadData = myReadData + c; but that resulted in it just saying"Error compiling". So far that seems to be what I get no matter how I try to re-write that line. |
| 22:06.35 | DarthExpeditor | sorry, wrong window |
| 22:08.27 | lorsungcu | ill say |
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| 22:11.53 | AlexForster | developers: other than call transfer, what are the scenarios where asterisk will masquerade a channel? I'm pushing a few thousand calls a day through an instance of asterisk 1.8, and 3-4 times each day, for only a few specific people, asterisk ends both legs of the call immediately after masquerading a channel as "AsyncGoto/{channel}<ZOMBIE>"... I'm running a mostly stock install of FreePBX 2.10, a |
| 22:11.53 | AlexForster | nd I don't have a single instance of AsyncGoto in my dialplan |
| 22:13.07 | AlexForster | alternatively, does anyone have any tips on debugging this (keeping in mind that it's a prod system with high volume, so I'm trying to avoid just "logging everything").. the SIP trace doesn't give any helpful info, nor do any manager events I've seen |
| 22:13.31 | jmetro | dont run prebuilt pbx systems |
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